What SAMPLE RATE Should You Record At? | Why HIGHER Can Be WORSE!

Why recording at 96/192Khz can sound WORSE than recording at 44.1/48, and why audio sample rate bears no relationship to frame rate in video. We bust some myths in an easy-to-digest video that explains the science behind sample rates, aliasing, and bit rate - as well as Mark playing with his new 55" iPad pro!
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[@officialyourdad2342]

Learned more from you in a half hour than from my university audio engineering courses tried to explain over a week. Well explained and thanks for showing examples. People like me learn from seeing and hearing examples not just having definitions thrown at them 🤘🏼

[@Producelikeapro]

Another fantastic video Mark! Marvellous work!

[@PresentDayProduction]

Thank you my friend, much respect!

[@iainparamor1482]

Warren are you planning to also get a robot in your studio that calls you names? :)

[@Producelikeapro]

@@iainparamor1482 Eric does that for free!

[@ximre2]

You can have these Audio Files if you want to see what we're talking about...THERE IS A DIFFERENCE THAT ANYONE CAN HEAR!!!

[@Electricowlworks]

@@Producelikeapro RoboEric, haha.

[@Pericles777]

The ONLY reason I use higher sample rate is for audio stretching. And it’s a rare occasion. Usually some sort of flown in sample that I want to stretch.

[@BlakeMlungisi]

Same here. Time stretching at 48kHz is okay if you have very good plugins like X-form for Pro Tools but 96kHz with standard plugins works much better for me. I do audio stretching all the time because I work with vocals groups that aren't necessarily professional.

[@nickager2008]

I record audio for film and tv and have been asked to record sound effects such as explosions at higher sample rates as post production want the option for time stretching with no artefacts. But still record 99% of our work at 24/48

[@dwindeyer]

As in time stretching while retaining the pitch? How does a higher sample rate result in less artefacts in that situation?

[@mwdiers]

@@dwindeyer Less interpolation.

[@ovonisamja8024]

@@dwindeyer Not retaining the pitch.

[@codyrap95]

That's what I expected from the video to explain. Pretty disappointing.

[@AnnaVannieuwenhuyse]

@@codyrap95 the distortion you get from exceeding nyquist itself is less important than the fact most recording converters just really suck at high sampling rates. I prefer 1080p 60fps over 4k 15fps. There is a sacrifice somewhere

[@5fiveyearmission5]

I have a music technology degree but have forgotten so much as the years have passed. Your channel is top notch!

[@RecordingStudio9]

I also record and mix at 48/24. The only reason I would record at a higher rate of 96khz is that my audio interface latency drops down to less than 3ms round trip, allowing me to use VST effects during recording with virtually no latency. Maybe that is a topic you can take on next?

[@g.m.6417]

Depending on what daw your using you can fix the latency even better for recording & play back. If you have an option to set the i/o buffer to a higher value for both. This is not to be confused with the buffer latency slider etc…

[@RecordingStudio9]

@@g.m.6417 Not sure I get what you are referencing. And, how a higher buffer size, in the driver or within your DAW can reduce latency. Think of it this way. If you have boxes lined up and at one end someone fills them with papers and push them to the other side. More boxes (higher buffer size) means it will take longer before the first full box reaches the other end for the papers to be taken out, hence introduce latency. This also depends on how quickly a box is filled and emptied on the other end (CPU power). If only a few boxes, then the CPU may not have enough speed to empty all papers in time for the next full box arriving (crackles and pops). Hope this simple analogy will help you out.

[@chrisfeatherstone9691]

In Electrical Engineering school we learn all about this. Everything you said was spot on. Nice work!

[@Whiteseastudio]

Excelent video! I agree fully that ADDA is always the most comfortable on 48kHz... However, I would like to see your take on the processing sample rates. I've experimented a lot with this (oversampling in the plugins etc.), and from my perspective it does make a lot of sense to have those parts of your chain running in higher sample rates.

[@PresentDayProduction]

We may do a dedicated video for this in particular, however, yes, oversampling in plugins is definitely useful. We found it sounded great on the FabFilter plugins we tested: kzbin.info/www/bejne/mJjUiKqAaNCcppY . We'll look into oversampling into plugins in particular in a future video. Thanks for your comment!

[@eliaszerano3510]

whats wrong with 44.1 ?

[@caseykittel]

@@eliaszerano3510 sure you pass the nyquist test in human hearing, but you get less artifacts from the plugins at the higher 48k 24-bit.

[@kyron42]

@@eliaszerano3510 there's nothing wrong with 44.1 if you're recording distorted guitars and drums.

[@eliaszerano3510]

@@caseykittel how about 88.2 then ?

[@duncanmcneill7088]

When doing the 48k vs 96k null test, it would be useful to have a spectrum analyser (Voxengo SPAN is my goto) to show where in the frequency domain the additional harmonics are occurring. Testing plugins for how well their oversampling algorithms work can be quite revealing - some are awful and the aliasing becomes readily apparent on loud high frequency content (particularly sustained tonal sources like Glockenspiel, Celeste and Triangle etc). Non-linear processing in the digital domain has definitely improved a lot with higher processor speeds allowing for higher oversampling rates.

[@ryanmaroney4793]

Best explanation I've ever seen on this topic.

[@PresentDayProduction]

Thanks, we’re glad you found it useful, it’s a tough topic to cover!

[@FlockofAngels]

Great video, lots of interesting info. I record vocals and other instruments at 32-bit so I don't have to set my levels at all and I never distort a take. I can sing very loud and very soft and never have to worry about my input level and where it is. The files are bigger but I have yet to have a project take up 1gb of audio data. That is with lots of vocal tracks and takes. And my 18TB drives are accommodating to these levels of data. Also, the 32-bit sounds better to my ears.😊

[@m9shamalan]

but your converters are still 24 bit, so you can still definitely clip them..

[@ibleasse]

I mostly use 48k at 24 bit. There are instances where I either need higher rates or higher bits. I use higher rates for audio restoration when digitizing and restoration work. It allows me to see all the problems clearly when I have to zoom right in. But it has no effect on sound. For bits higher than 24 on the other hand, I only use to record sound in the field (sound effects). I again, only use it rarely when recording extremely loud sounds (gun shots, cannons, explosives, jet engines, rockets, Formula 1). Higher rates and bits, I assume are also useful for scientists who use sound and acoustics in their studies (bioacoustics, stellar acoustics). But for musical purposes, 48k@24 is perfect.

[@IsaacJDean]

Can the AD/DA converters you use actually handle the extra dynamic range though (for the examples you gave of reasons to use higher bit depth)?

[@ibleasse]

@@IsaacJDean Steinberg AXR4U can handle it

[@aristosxanthus514]

Underrated Channel. Great editing and explanation of concepts with the graphics. Keep it up!

[@PresentDayProduction]

Thank you, Aristos!

[@txtrader512]

should render out the difference between the 192k and 44k versions and pitch shift it down. would be interesting to see what that sounds like.

[@nexusobserve]

Pitch shifting is interesting for monitoring, it gives you time to think. We've all press halftime in Pro tools.

[@seybsnilksz]

Hey guys, mr. Know-it-all here. I've really been enjoying your videos since finding them today. Very informative, very well explained, and the British accent is superior :) I did disagree with one thing here though where you said that no dither is required when exporting at 24 bit. Since most (all?) DAWs operate internally at 32 or 64 bit floating point, the bit depth is indeed lowered when exporting something to even 24 bit. Of course it's debatable if it's ever audible, but truncation distortion is in fact introduced. And in my opinion, a bit of inaudible dither is better than potential unwanted errors in the audio. Again, love your videos and it is one of my dreams to work with a team of such nice people some day! :)

[@PresentDayProduction]

Thanks for your great comment! You’re right, but as you said, if it’s audible is another question in itself! Glad you enjoy our videos, this one is one of our favourites!

[@seybsnilksz]

@ReaktorLeak You mean that the noise from any component during recording will be acting as dither? That wouldn't work since the noise is printed and "frozen" with the file when recording and is then no longer random. Hence why dither would be needed when rendering/exporting a new file with the material.

[@seybsnilksz]

@ReaktorLeak When it's recorded, it's printed into the file and then it's the same every time.

[@sekritskworl-sekrit_studios]

My FIRST video with you... and you ALREADY SOLVE one of my biggest curiosities as I am a NOOB to Audio. Thank you. And, Happy Holidays!

[@PresentDayProduction]

Thanks for watching! Glad we could help :)

[@berndkiltz]

Wow. 30 Years recording digital and now I understand it. Kudos to you!

[@Mrspkey]

Thank you, thank you, thank you. Despite having a degree in audio tech and have been recording my own music for the last 25 years, I became almost fixated with the idea of more is better. As a result, I always struggled to optimise my setup to match the continuously evolving marketing requirements and pressure from peers. I think you have just demonstrated in layman's terms the reason why going as high as 96K, 24bit is not only unnecessary but potentially, even harmful. I am now hoping that going back to 48K, 24bit will give my computer another couple of productive years whilst I'll also perhaps enjoy the process more now I won't need to keep optimising my setup for unreasonable specs.

[@RS-pp7ng]

This is easily THE greatest explanation on this subject. Ever - and with a great dose of hilarious British humor! Thank you so much guys, we're forever grateful for this. Bless you all.

[@PresentDayProduction]

Thanks R S! Thanks for your comment, I’m glad it was helpful!

[@Deluxeta]

I am using 96KHz for editing purposes when I'm capturing audio or working on a project within a DAW. It's much easier to varispeed a 96K file at extreme percentages without it sounding off. As for capturing audio, it's much easier to edit out crackles and pops on a needledrop or cassette recording than it would be at 48KHz. When it comes to getting stuff out of the box, I always use 48K. Home at Last sounds fantastic regardless of the sample rate.

[@mrnelsonius5631]

There’s so much misinformation about this subject! Thank you for a great video. Also: just because a plugin offers oversampling options doesn’t mean higher is better. The quality of oversampling can vary and often sound worse than defaults. Always trust your ears, not the numbers :)

[@PresentDayProduction]

Yes, that’s a really good point! And ‘always trust your ears, not the numbers’ is so very true. Thanks for your comment!

[@nayaleezy]

6:40 that 96khz beep had so much more warmth & presence, was it recorded on an Neve 1073 preamp?

[@PresentDayProduction]

Come on Lee, it was an API! You should have heard that! 🤪 The transients!!!

[@gareth432]

@@PresentDayProduction even though you should have compressed the beep to glue it together ;-)

[@d.lafollette]

@@gareth432 It's all about the beep glue.

[@ruidanin_rocker]

Ahahahah! Very good!

[@mthomas1091]

Omg like butter!

[@Chrisreedbeats]

I love the comparison between audio and video!! This made both concepts click for me, as I enjoy videography and music production. Thank you 🙏🏾

[@AnonymSuperhero]

Wow so much useful information! Especially the AES Test Tone part to put it into perspective. I did this test myself and I'm shocked that I can clearly hear the unwanted noise in my monitors at 96+kHz for both test tones. Considering I use a typically well regarded audio interface around 250€ this alone definitely convinced me to stay with 48kHz for my recording, mixing and mastering. Thank you for this! I'll dive deeper into this topic and I'm looking forward to more videos :) Best regards from Germany

[@dmc5747]

Thanks for the video, and thank you for the invitation to disagree in the comments (as long as we include some supporting evidence). I totally agree that 16 bits at 44.1k is perfectly good for music reproduction. However, I have twice recently NEEDED to record at 88.2k. Both times it was because Fiverr session violinists had sent me recordings that were unusable due to their recording hardware not coping with the surprisingly high amount of ultrasonic energy from their violins. The first session player was using their computer's onboard audio input, and after I heard it I viewed its spectrogram to see what was causing the harsh sound. The aliasing was clearly visible because the violinist's vibrato made the harmonics wiggle quite a lot, and the aliased wiggles were all upside-down. I asked for a revision recorded at 88.2k and then I down-sampled the file to 44.1k on Pro Tools so I could import it into my session. This removed the ultrasonic frequencies digitally and left me with a surprisingly good recording. The second session musician claimed to be using a Focusrite Scarlett 2i2 (I have no way of knowing for sure) and their recording displayed the same upside-down vibrato when recording at 44.1k. So, the moral of the story is that if the audio interface has not got what it takes to filter out ultrasonic energy, then it is better to record at 88.1k or 96k so that the Nyquist frequency is raised above that energy. Afterwards, you can rely upon digital filtering to remove it while converting to a sensible sample-rate.

[@Pipelyd]

Ultra sonics are definitely generating some extreme Steely Dan sounds in this track :) Thanks for the well explained tech. stuff.

[@coisasnatv]

Using the same principle of Nyquist, isn't better to record at 24bit 96kHz and export the final project as 24bit 48kHz?

[@sicknoterecordings6909]

Nice to see this hammered home with some excellent illustrations and science to back it up. I just did my first project in 96k as an experiment and my system just about coped. Its funny how your mind convinces you it sounds better. But maybe I'm just getting better at production and mixing? 🤞

[@BigHugeYES]

Could be placebo, but I hear 96k responding better to elastic audio, pitch correction, and some plugins.

[@PresentDayProduction]

Higher sample rates can help with some time stretching software, and non linear plugins that don’t oversample (or do but don’t have it turned on!)

[@freakkyt]

One factor that would benefit from higher sampling rate is I/O latency, especially for those who rely on non-hardware monitoring while recording. A common example are guitar players recording through IR speaker simulators in the DAW. As long as CPU and HW resources in general allow it, 96Khz usually guarantees the 5ms or less latency most player need to get a realistic response during a recording session, without having to lower the buffer too much and incur in stuttering or artefacts. Apart from this potential reason for higher sampling rate (which is specific to recording rather than distribution), all the considerations in the video are agreeable, especially since you pointed out the internal oversampling often done by plugins dealing with high harmonic content, such as compression and saturation DSPs. Thanks for the vid, keep up the good work!

[@minimalmayhem]

Yes... I've found that working at the higher sample rate makes my studio latency much snappier. Should I then be individually re-sampling these files recorded with higher sample rates to the lower rate, or will they sort themselves out when I change the whole project down to 48K for the final mix down?

[@ezrashanti]

@@minimalmayhem Not worth it. 96k is good. The fact that it can record frequencies up to 48 K is actually an advantage. There will not be aliasing of the frequencies between 24K and 48k as they are recorded accurately. Aliasing occurs when harmonics above the Nyquist frequency bounce back down below it.

[@freakkyt]

​@@minimalmayhem That depends a lot on HW resources. Personally, I do simple stuff for home use, nothing pro or commercial, so working at 96KHz and exporting the mixdown at 48Khz works fine. Some DAWs (Cubase for sure, don't remember Logic) prompt you asking if you want to change the sample rate of all recorded samples when changing the project sample rate after recording, even asking if you want to move or keep samples at their location, so if you want to record at 96+ for low latency, but prefer to mix at 48Khz togo light on CPU/Mem/Storage, that's totally doable. The main limitation is that if want to record new parts after the samplerate switch, you'll have to do it on the new "mixing" setting, with 48Khz and probably lots of plugins loaded, so the I/O latency will be higher.

[@freakkyt]

@@Indrid-Cold Interesting video, but we need to be careful when looking at research data, as it's very easy to be deceived into deducing incorrect absolutes. The capitalisation in "FASTEST transit rate and ANY nerve impulse" appears to suggest that it's humanly impossible to perceive audio latency under 80ms, and we all know that's not the case. Stimuli perception has been studied in depth and there's plenty of interesting literature on it. Perception has been found to be different for visual stimuli (the 80ms Vsauce is talking about, some studies estimate it more around 50ms), touch (approx 50ms with that interesting nose vs toe delta explained in the video) and audio stimuli (approx 10ms), with tests showing substantial differences between casual listeners and musicians. Those times are always averages, not physical constants. Back to audio latency, let's also keep in mind that the I/O latency indicated in the DAW is pure processing time (bufferSize/sampleRate=512/48k=10.67ms), which is only the portion of total perceived latency contributed by DSP. In our machines there's additional buffering and delay happening at drivers layer (interrupts in audio and USB drivers) and can contribute an additional ~10ms, or as low 3ms with latest USB C drivers at 96KHz (again, latency improves with higher sample rate).

[@mikemckernan1076]

@@Indrid-Cold I think you may be mixing up m/s (a speed) and ms (a unit of time).

[@rockstarjazzcat]

Nice use of the video analogs to illustrate the bad assumptions! Kind regards, Daniel

[@justlooking813]

I use the higher sample rates with my Sanken CO-100k and hydrophones for capturing sound effects to be played back at 48K. Pretty niche use case, but absolutely necessary for high frequency capture.

[@petrub27]

So you pitch down few octaves?

[@justlooking813]

@@petrub27 yes!

[@phiprion]

24/96 is great when you need to pitch/tune vocals or samples; Analog emulation plugins give a much better result espicially at higher freqs (anything that brings saturation), it's very clear with synths or cymbals or overdriven sounds

[@antigen4]

Actually it’s in the low frequencies that we separate the wheat from the chaff with converter quality and where the higher sample rate converters really shine

[@Wizardofgosz]

@@antigen4 Explain.

[@antigen4]

@@Wizardofgosz better converters tend to have the BIG payoff in the LOW frequencies at least as much as in the highs ... what more is there to explain?

[@Wizardofgosz]

@@antigen4 let's see some science on this please. I call total BS. Since lower frequencies are not the hard frequencies to reproduce. Or is this just religion?

[@antigen4]

​@@Wizardofgosz - gotta get up PRETTY EARLY in the morning to fool YOU huh?? haha ... ok don't take my word for it - you wanted the explanation. go listen to some proper converters sometime and let's see what you think. go demo a BURL of Forsell or something in the 20-30K range and compare to an inexpensive soundcard or what have you. this is exactly what i was talking about when i mentioned (in another comment) people dwelling on the superficial ...

[@letsallbe-friends1120]

*I feel like I'm watching the audio engineer equivalent of "Red Dwarf" (I love RD BTW!😏)* 😄👌

[@PresentDayProduction]

Letsall Be-Friends Funny you should say that just as I’m putting the finishing touches on mastering the new Craig Charles (Dave Lister) Trunk Of Funk album! 😎

[@letsallbe-friends1120]

@@PresentDayProduction Wow! That's amazing! His character was the mentor for all my adult slovenliness! 😅😅 I'll definitely keep and ear out for the release. 🔉🎶👂🕺✨ Fantastic channel BTW! 🙌🙌🙌

[@TerryMaplePoco]

I also got the RD vibe and I'm thrilled about it

[@TerryMaplePoco]

@@PresentDayProduction amazing!

[@DavidHamby-ORF-48]

Mark does a brilliant job with this topic. Everything he says squares with the work of Shannon, Nyquist, et al. Mark explains the potential harms of oversampling (faster than at the Nyquist rate).

[@Yupppi]

Great stuff. I had already forgotten quantizing, AD/DA conversion and Nyquist and was in the state of "just knowing". This brought back that in simple and easily understandable terms. On top of human hearing only reaching about 20 Hz to 20 kHz, people who the most argue about that stuff are the ones whose hearing often doesn't reach higher than 12 or 14 kHz either, leaving them out of the paradise they're so keen to protect. Not only that, it often feels like many sounds at the top end are nasty unless it's just some echoey shimmery whoosh happening there making the sound and space feel bigger. And many producers love old analog gear like boards that have a sweet top and low end cut. Even better, the mixes often sound better when you cut some of that outer limit of hearing range and beyond rumble and shizzle, making it sound more powerful and focused without adding anything and seemingly not removing anything (a lot of what I said was of course said in the video as well and I seem silly stating it again). Also it was some producer on youtube that showed a plugin that adds tape saturation like distortion to the mastering, might as well use that to get a bit of that warm grit, since it's designed with that goal in mind. Guitarists also often like to criticize limited bandwiths of old digital gear, and at the same time love using such a tape machine that has a worn out tape and a preamp that heavily cuts high frequencies for smoother top end tone. Weird bunch. I just recently acquired a nice piece of 80's rack effect, a Yamaha SPX90, which had only range up to about 12 kHz and it didn't even have proper AD converter, but a combination of chips doing some kind of trick to digitize the analog signal. The result was a pretty nice sound just because the signal goes through the unit even without effects, even though it is digitized and loses some high frequencies. Just like Echoplex's preamp was used without the tape delay itself, just because it sounded nice. An example of how it is often the case that technical limitations provide happy accidents. Dither is another funny thing in video context. And gaming context. Playstation 1 had crazy amounts of dithering to make the games look great while saving in technology costs and processing power, allowing more complicated and fantastic games and graphics to exist. The mention of fps in movies and tv made me remember how some modern shows look weird and unnatural for the technological advancement of increased recorded framerate. I've been looking for a camera that can do video recording as well, and just now realized 60 fps might not even be what I wanted for filming myself. Thanks!

[@gammakeraulophon]

Great in depth video.. thankyou.. There's only a couple of areas you neglected to address, such being the use of higher sampling rates where audio recorded is intended to be repitched in music composition.. once such audio is pitched down by 2 octaves then surely theoretically the stuff we don't want which is floating outside the upper limits of human hearing can be brought down into the audible spectrum.. thus making higher sampling rates more effective at keeping the audible band clear if sound is destined to be mapped across a keyboard of several octave range. Also though.. as I understand it.. high sample rates are used in such pursuits as Oceanographic Audio Research... But for the same reason.. that there are sounds captured which are of interest to the scientific research, but which lay outside the human audible band.. whereby such sounds are again destined to be repitched in order to bring them into audibility.. By oversampling at 96 or 192kHz, one can again repitch downward by up to 2 octaves without bringing the sampling frequency itself down into the audible spectrum.

[@weschilton]

People keep saying this nonsense and I have yest to see one single example of this being done anywhere. How can this method even be practical?? How exactly do you use these mythical ultrasonics? I mean you can't HEAR them so how do you even know what you're getting or even IF you are getting when you pitch them down?

[@gammakeraulophon]

@@weschilton Is not nonsense.. is marine science.. If marine science is using it then it is to some intelligent and practical purpose. Just because you yourself do not understand something.. or it is outside of your sphere... does not merely make that something 'nonsense'. You only present yourself as ignorant. As for everything I said about recording samples for intended repitching in musical composition. Such is patently true. It is important to keep the clock signal out of the audio band. Either you undestand a subject or you do not. Takes no effort at all to call something nonsense, merely because you cannot be bothered to make some effort towards understanding. Casual and lazy.

[@keithspillett5298]

Fascinating stuff beautifully presented. I can 'perceive' a difference between 24/96 and 24/48, but have never been quite sure why until I watched your video! I'll definitely standardise on 24/48 now. I've always used it for video soundtrack work, but will now also use it for my CD output 🙂

[@2112jonr]

Would be interesting to do it in a controlled bind test.

[@PresentDayProduction]

44.1K is most suited for CD, rather than 48K. Better to work with - and export in - what you need; rather than having to get it converted when it’s then burned onto CD

[@keithspillett5298]

@@PresentDayProduction my CD authoring software automatically converts whatever is thrown at it, and auto-optimises accordingly. Much of my output these days is supplied on DVDs at 24/48 anyway.

[@valoelios40]

According to Ian Shepherd, dithering actually matters also at when bouncing audio to 24 bits. He points out that audio in a DAW is processed at s higher bitrate (32 bit floating point) than 24 bits, eventhough the material was recorded at 24 bits. Henceforth dithering would be a necessity also when bouncing audio to 24 bits. Please correct me if this is not correct.

[@GuidoRighi]

you are right

[@johnc.8298]

I am 100% in agreement with your summation of 48/24 as this also circumvents any distortion and coloring issues due to the addition of noise that is introduced by the dithering process to a lower sampling rate. Why choose this distortion? One additional consideration I recommend is to decide before you begin the project whether the final product is destined for video/on line distribution or to create a CD. If a CD, then sample at 44.1/24. If video, 48/24. Avoid dithering, the process deliberately introduces noise to fill in "holes/gaps" at the lower sampling rate.

[@rajeshnair4399]

Hi. I’ve been wondering how you played back files of different sample rates from the same logic project. Am I missing something here?

[@PresentDayProduction]

With GREAT difficulty... in the end we upsampled the 44.1 and 48 to 96

[@EpithetMusicTV]

@@PresentDayProduction what do you use for SRC?

[@javytorres94]

This video is free?! My God the information! Thank you guys! Great content! New Sub!

[@DannyTaddei]

I’ve been working in film and music recording my entire life. I’m 58. I absolutely agree with your assessment and thank you for the simple proof of it. Many people need to see this. You did a great job of making it simple.

[@PresentDayProduction]

Thanks for your comment Danny!

[@jye.acoustic]

I use a focusrite 212 audio interface via a mixer to my phone via a usb adaptor. One thing i have noticed is the focusrite 212 operates at 192 24bit.. fine for recording.. but unfortunately can't adjust the sample rate of the focusrite.. although it is deemed class compliant, i have noticed the differences in social platforms when live streaming.. for example - live streaming to youtube the sound is as you say distorted somewhat.. yet live streaming on tik tok its much better.. live streaming to facebook i have found changes dependent upon their latest updates.. which is frequently. This video is very useful in helping me understand whats going on.. and i have to agree that lower would be better for audio. Trying to figure out now how i could improve my audio between online platforms without using plug ins or Daw etc... as trying to keep my set up minimalist. Hope you could do a video with regards the above for best live streaming audio set ups.. in relation to different social platform requirements & using audio interfaces would be very helpful. Thankyou for the post. Very helpful. 🤗👍

[@timnordberg7204]

Finally, a definitive answer--that merits a sub. Thanks for the great video. I do have one use-case-specific question in mind: if I were to record a sound (let's suppose the mooing of a cow) with the intention of playback at 0.25 speed (think sound design, not music) would this be a case where recording at 192k would be the preferred method--given that the sound will still have 48,000 samples in each second after it's been stretched to quadruple length?

[@PresentDayProduction]

Is the cow mooing at 96khz? Are there bats in the vacinity that need to be captured alongside the cow? If not, then you don’t need a higher sample rate. If the cow is mooing at, say, 2khz and you slow that down to 0.25, it’s now mooing at 500hz. So a higher sample rate is only useful if there is information higher up in the frequency range you wish to capture. Will you get audible artefacts if you record it at 48khz and slow it down to 0.25? No, because you only need a sample rate of 0.25 - 12khz - to play that back. Hope that makes sense!

[@bigmacmillerlite8775]

If you guys keep this up then you will be Kingz of teh internetz. Excellent channel Bong Friends.

[@PresentDayProduction]

Thanks Big Mac! We’ll try our best to keep it up ❤️❤️

[@simongore29]

This is the best audio production channel on KZbin. Thank you.

[@PresentDayProduction]

Thanks Simon!

[@nachtaktiv]

i'm using 44.1/24. t.m.k. spotify also uses 44.1 and almost every sample that i have is 44.1. i don't want to convert every sample that i'm importing. in some DAWs (like ableton) you don't notice it, but in cubase there's a convertion that takes a few seconds and it generates a new file with the new sample rate. and i think, upscaling ends not in better quality.

[@derChili]

This is by far the best video of the subject I ever saw. Great job.

[@PresentDayProduction]

Many thanks Towi - thanks for watching!

[@reverendcarter]

pitch shifting is much smoother at higher latencies. halving the pitch of a 96k file still means its at 48k resolution so its not a grainy and artificial sounding. also the latency at the same buffer size is lower. but i do agree about everything else.

[@peetiegonzalez1845]

Yes. As an "audiophile" I'm perfectly happy with 16/44, but as a DJ I would much rather have a much higher resolution so I can pitch-shift and beat match without it sounding weird or introducing artifacts.

[@mwdiers]

Yes, in mastering and mixing there is often an advantage at working at 96Khz when you are doing things like pitch shifting, etc., particularly when using plugins that do not do oversampling (Soundtoys, I'm looking at you). But upsampling a 48 or 44.1 file to 96 will give you the same advantage without the risk of distortion, because of the intrinsic bandwidth limit. Live DJing with such effects is essentially equivalent to mixing ITB. In a final master, though, it makes no sense to distribute anything over 48.

[@olegoleg1838]

96 does sound richer. i don't know how but i watched a video of a blind test here on youtube and when you start hearing those with higher sample rate the effect is like you hear the real thing more and more as opposed to a "reproduction". it is actually a very satisfying feeling

[@gordongurley3982]

@ReaktorLeak that demo is only testing playback. I’d argue that when recording music, higher sample rates are beneficial, the most basic reason being that there is less latency through the system.

[@gordongurley3982]

@ReaktorLeak Maybe not you, but PDP video is arguing that there is no reason to record at anything higher than 44.1 or 48k.

[@myproductionadvice]

96 and 192 do sound much better on a poor clock audio devices as the faster speed rates would decrease the jitter effect.

[@petrub27]

It doesn't. The sound wave was already reproduced correctly.

[@Simbabhebhe]

Straight be coming one of my favourite Audio tutorial channels along with Produce like a Pro. This is hands down the best video in Sample and bit rate I’ve EVER seen. It’s crazy how much misinformation out this and this has really cleared it up for me and given me confidence before my first paid on location recording that’s coming up. Top works gents (and Grace of course :-)

[@PresentDayProduction]

Thank you for the kind words Simba! Glad you found it helpful

[@Wawisupreme]

As an audio engineer myself, I should point that most of what you said is true, nevertheless, there is some utility in high sample rate recording, specially in audio design and postproduction. Being so much above nyquist gives pitch headroom to manipulate (lower by several octaves depending the sample rate) audio without losing detail. In that regard ultrasonic information is usefull.

[@Wizardofgosz]

Ummm, WHAT?

[@Wawisupreme]

@@Wizardofgosz it is a bit tricky to explain, but if you record something @ 48kHz and then you pitch that audio file an octave lower (but in a traditional way, not with a process that adds interpolation like some plugins do) you are basically dividing the number of samples in half, and doubling the duration of the audio file. By doing that the real sample rate of the resulting file would be equivalent to 24kHz, which means that you are now not able to reproduce audio information in the whole audible spectrum do to the Nyquist Theorem, the half of which is the highest frequency you can accurately represent ( in this case 12 kHz). If you record at higher sample rates, even the crazy 384kHz, you should be able to pitch down several octaves before running out of pitch headroom. Some audio applications could benefit from this fexibiliy.

[@weschilton]

@@Wawisupreme Did you seriously just say "pitch headroom"??? hahaha!

[@natdenchfield8061]

What's not to understand? It's a great analogical phrase that simply describes what they are talking about.

[@danield2000]

It all comes down to one thing, what sample rate is the final master going to be? Any conversion in digital audio is less than ideal and you should record at the same rate you are going to playback.

[@rdoursenaud]

While this was true in the early ages of digital audio, this is not really the case anymore. Sample rate converters have come a long way and produce distinguishable results now.

[@vleevision7787]

THANK YOU!!! been saying this for years!!! it comes down to your DAW's specs and YOUR EARS!!! 24/ 48 is awesome for me . . .

[@m1ke1981]

I'm loving this channel! Just discovered it some hours ago.

[@austinbuttenob4849]

As a video guy, the comparisons between higher sample rate and frame rate was great!

[@PresentDayProduction]

Cool, thanks Austin!

[@JeremyHalterman]

Is a higher sample rate / bit depth beneficial for tempo stretching? The shorter length and higher dynamic range could theoretically maintain the smoothness of the audio as it’s stretched, but if the DAW oversamples prior to the stretch, the outcome should be the same-? I might be putting my ignorance on display here 🤪

[@PresentDayProduction]

It depends on what you’re doing - theoretically, yes, and theoretically, no! The best way is to experiment with different sample rates in your own projects and see what works/sounds best to you. I’ve time stretched vocals 10-15% for remixes at 44.1 or 48khz, and there have been artefacts, but not because of the sample rate, because I should have re-recorded the vocal at the new tempo!

[@marvinrockon]

2. video i watched, just discovered your channel this day. Subbed, Bell - undebatable great stuff and i love it. Interesting, scientific, entertaining. Criminally undersubbed!

[@Sokolemon]

Great video, and excellent explanation of the psychoacoustic's theory involved. While I agree on the whole with this, I would also comment that given the right equipment, and the right listener, Higher sample rates have always sounded more like the 'glue' is working better. A sine wave is only one element, and can not interact with other sounds. Higher resolutions always (to me) had better 'air' and richer bottom end. Just overall better clarity IMO. but this is comparing 16/44.1 to higher rates. My gear (lol, and most likely my ears) doesn't seem to produce any more clarity from 48 upwards. I am sure there is a difference in the air and bottom with the right gear (and ears ;). Cheers.

[@gabriel_kyne]

great! Can't believe you only have 14k followers, it's like a professionally produced television show!

[@PresentDayProduction]

Thanks Gabriel! Thanks for your support, we are working hard on some great new content for next year!

[@jayddd4946]

Interesting, but something I don't understand. You said this song was recorded at 24/96. What exactly do you mean? All individual instruments and vocal sources were recorded at 24/96? Then, you re-recorded each track identically in a new project at 24/48 and then at 16/44? Obviously not, thats impossible. So could you verify what the original source was recorded at, how exactly the variations were created, and if the analog track sources contained information above 20K? Also, I think recording 30 tracks of stuff at 24/96, and recording 30 tracks of 16/44 would yield different results when combined, compared to a 24/96 stereo mix being re-recorded or downsampled. And you don't hear digital information in a Logic file - you hear what is output through the D/A converters, amplifiers and speakers, though the air into you ears, then your brain processes it. I think there are quite a few more variables in this than it might seem. Is recording at 24/48 good enough, yep, for sure. But insinuating that it's pointless to use anything above that, as a general statement, is not accurate. (with respect, love your channel, and correct me if I'm misunderstanding anything).

[@PresentDayProduction]

Hi Jay, the song was originally recorded and mixed at 96K. We then took all the plugins off, converted the individual mix channel files to the lower sample rates and mixed it again at the lower rates using the same plugins on the same settings. And yes, you hear the output through your converters and amplifiers and speakers, which are generally designed to reproduce 20hz-20khz or thereabouts. Now I didn’t hear any difference on my system between the lower and higher rates, but I’ve got £30K invested in a playback system - a lot of people have commented that on cheaper monitoring systems they can hear a difference. And that’s my main issue with higher sample rates. If your electronics are making it sound ‘better’ or more ‘analogue’ then that’s not good - because that isn’t happening in Logic, it’s happening in your monitoring system, and only you are listening back on that! That’s why we included the test tones. And that’s why a lot of people try 96 or 192 because they assume ‘higher is better’ and hear a difference when, in actual fact, for them, on their system, higher is worse.

[@fuglbird]

@@PresentDayProduction So you are actually not comparing three recordings of an analogue signal with different sample rates. You are just comparing correctly down-sampled versions of the same digitized signal, which in my opinion makes the entire comparison invalid. Try sending the sum of two sine waves to a speaker - one 9.5 kHz and one 19 kHz. Make sure that the sound pressure are the same from both frequencies. You may need to adjust the signal to the speaker a couple of times. The purpose is to test the phase at frequencies close to half the Nyquist frequency. Now add a third sine wave at 28.5 kHz. The amplitude of this frequency is of no importance. The purpose is only to check the aliasing filter. Finally take the recordings at your three sample rates and compare the phase of the recordings - or just do the subtraction again. This a very crude and simple test; but it will focus on the A/D part of the process.

[@sandernightingale]

@@fuglbird If you record at 44.1Khz no "third sine wave at 28.5Khz" is coming through. It's not in the signal, it doesn't get captured. What are you trying to prove? It's very very very simple. A 44.1Khz recording will be absolutely accurate up to 20Khz. You can prove this mathematically. The converters are also easy to make unlike 96Khz or higher ones. You cannot hear anything more. There is simply no reason as far as audio quality goes to go beyond 44Khz or 48Khz.

[@fuglbird]

@@sandernightingale Then prove it mathematically - here! In analog to digital conversion you make a lot of compromises. You need to avoid aliasing and you need to preserve your amplitude and phase properties up to 20kHz. You cannot do both 100% when you are sampling at 44.1kHz. Please do two things: 1. Show your mathematical prove here. 2. Try testing it yourself with a simple tone generator. There is one reason to go above 44 kHz and that is to get better anti aliasing filter response below 20 kHz. Off course you will down-sample to 44.1 kHz or 48 kHz. Of course we cannot hear the tones above 20KHz; but we can reduce the distortion of the signal below 20 kHz caused by the anti aliasing filter.

[@berkut6313]

Very well put video and explanations. Had watched the fab filter too. Have one concern though (although I’m perfectly happy with 48/24), it is that Nyquist theorem applies to periodic signals. Fourier says any complex periodic signal can be decomposed into an infinity of sine waves (but only the fundamental and harmonics in the hearing range need to be taken into account, which rightfully caps the n harmonics to 20-ish kilohertz and therefore the sampling rate to about twice that). The only problem here is that only EDM targeted at Goldfishes can be seen to fulfill the criteria of both Fourier and therefore Nyquist. The very nature of a transient signal (plucking of strings, voice attacks etc) is that you can’t tell much about it in the frequency domain, and therefore the whole mathematics tumbles down because you’re outside of the applicability range and conditions. Still, low-pass filters characteristics are doing a pretty good job at filtering ultrasonics without killing details, and they are known to us in the frequency domain only (gain and phase Bode plots, are equivalent to a frequency sweep). When in need to have a feel for what’s going on in the frequency domain for a non-periodic signal, you can use what is called Power-spectrum density (PSD), wich relates energy content (per time unit hence power) to (pseudo ?) frequencies. But then, it’s only a limited overview and time-response (f(t)) is pretty much all you are left with. In audio, this is you hearing the thing. So there might be a rationale to push for higher sample rates to get more subtle details on highly resolving systems, although my hi-fi, room and listening conditions didn’t quite allow me to hear a difference in such transients between CD quality and 192kHz versions of the same recording. I mean, I think I did (on Pianos, Acoustic Guitars playing classical music in isolation), but I cannot be sure this isn’t my brain tricking me into an “I want to believe” situation. Even though, It would not mean none can hear it on a better system with a better room and better conditions...

[@korayem]

23:00 That back to back comparison is brilliant

[@seattlegroovescene]

Excellent thanks for this video!

[@TarekTawakol]

I love the video, it's been a dilemma for me for so many years. I've settled to work 48khz/24bit for almost 5 years now. However, there is an aspect you are disregarding and as a mixing engineer i have experienced it first hand. Plugins- Plugins run algorithms. Algorithms processing higher numbers (sets of data inputed) yield higher results (or at least different ones). So running a vocal file recorded at 48khz through a seventh heaven reverb plugin Itb sounds very different than running a 96khz one. Processed reverb might sound smoother or more "lush" simply because the data coming out of it is higher. In photo editing for example, photos are finally viewed on a screen with a maximum size of like 30x25cm, prob jpeg compressed at 1280x720p. But in order to get a perfect edited photo, the original content was MUCH much bigger so editors can zoom in and craft details like eye lashes or pixel editing. The higher the resolution, the more control they get over altering the content. In higher sample rate sessions, especially with plugins processing, the story seem to be similar. Try it with simple waves rbass plugin on a kick, you will have more control with the higher sample rate. With multiple tracks, now you see a difference. Once you finally mix all that down to a 44.1 it just sounds so sharp and tight, just like that big res photo you edited down to jpeg... Try it out and lemme knw - also love from Egypt y'all!!

[@PresentDayProduction]

Thanks for your comment! Most plugins upsample for the reason - check out the excellent video from fabfilter, that explain that part much better than me! 😉 love right back at ya! ❤️

[@Fox_is_Fox]

As an "award winning" music producer I agree with you. There's a reason why high profile engineers work at high resolution. That aside, in my line of work music must be the center of attention, a good song overcomes any small technical fail so, to relieve my cpu I work at 24/44.1 ;-) Thanks for the input PDD!

[@rafaelvieiraprodutormusica3489]

Great video! I never used sample rate above 48 in any of my project only because every time I tested I could not hear the difference, even with colleagues swearing that they could (I always get a vibe of "The Emperor's New Clothes" when people talk about high sample rate). And also I always thought that the file sizes were not practical. Thanks for confirming that the little voices inside my brain were telling me the truth.

[@roberthedin909]

I seem to be settling on 24/48, before watching this video, for reasons mentioned in the video, and because I like having creative options, sometimes I like the sound at 48k for virtual instruments and plugins, and sometimes I like the sound at 96K when I use Metaplugin to x2 the sample rate. So, make things easier on the computer, and have more sound options.

[@firstnamesecondname5341]

🤔 20:25 but could the ultrasonics be issuing secret ‘instructions’ 😉 🤦🏻‍♂️

[@PresentDayProduction]

Damn, you’ve sussed it!! 😱 If people would stop recording at 192 it would be the end of the current pandemic too!

[@mrnelsonius5631]

Well, we know dogs can hear them.... and dogs eat poop. So maybe it’s a good thing we aren’t getting those instructions? ;)

[@2112jonr]

Only the cats can hear it. Messages from the home planet. Why do you think they're always hanging around in synth studios ;-)

[@DaskaiserreichNet78]

Would you make an exception in case of recording for a sound design where the plan is to time stretch the recorded sound in order to pitch it down by one or two octaves?

[@PresentDayProduction]

No, if you pitch 10khz down by two octaves it becomes 2.5khz, and you only require a lower sample rate to reproduce that. The only reason to go for higher would be if you actually want to pick up ultrasonics, or if it just ‘sounds better’ to you - that’s all that matters at the end of the day!

[@amdenis]

Great video. So well produced with so much accurate info. Thanks- you do the recording world a great service with videos like this. Question- what about recording DSD256 via appropriate mic’s like Sanken Chromatics of woodwind, vocals and strings? I’ve done null and related testing in our well isolated triple walled recording/live rooms here and have found some useful technical and creative advantages, which I assume you are well aware of. Obviously, we are talking about a much more expensive recording chain and construction standards than many studios can justify, but the differences are very interesting and quite different and very useful.

[@teashea1]

excellent content and style and production values

[@Limbiclesion]

Really good explanation of the quality issues related to various bit depth and sampling rates ...24 48 is certainly my way forward. 👍🦄🙏🎩

[@KariKauree]

In all explanations of audio sampling theory/the Nyquist Theorem that I've seen, a simple sine wave is always used. Would be nice to see it illustrated using a highly complex waveform for a change. I think it would put skeptics in a more accepting state of mind before going into all the other explanations and demonstrations.

[@JimRobinson-colors]

Being part of the online post production community, I find that there is always arguments about videos running out of sync. And the first comment is always : "what frequency rate did you record your audio?" They have this idea that 48khz mixed with other sample rates - i.e. 44.1khz ) makes the sync drift over time. I have jumped in and pointed out that frame rate has nothing to do with how many times a sound is sampled per second. A second is always a second. I do remember when digital audio started to creep into recording studios that we had to designate a master clock and set devices to be slaves, to stay in sync. So I am now going to save this share link and use this to support the explanation and discussion on sample rates in audio - Would love to hear a more educated explanation of audio drift ( sync ). Thank you for these detailed explanations of sampled audio.

[@robdm9838]

I would literally pay for content like this

[@tonigeiling]

I really like your video, I remember when the ADAT went to 18 bit recordings and later higher, we did A / B tests and the higher bit rate sounded better, at least we imagened. Today 24 bit for the headroom and the processing of rooms etc. suite me fine. I think the 96 ... khz options are there for marketing reasons only. Thanks for your well produced knowledge video.

[@KristianDowling]

You guys are incredible and I wish your channel every success. Thanks so much for the effort and quality of this video 🙏🏻

[@PresentDayProduction]

Thanks Kristian, we’re glad you enjoyed it! We’ll keep the content coming!

[@bobsandidge9613]

Just found your channel. You are both terrific. I appreciate the solid tech info as well as the pithy humor! I'm a long time audio guy and always learn something new from you. Thank you for the work you put into your very useful videos. I'm a fan! In Peace.. Bob

[@PresentDayProduction]

Thank you so much!

[@okay1904]

Excellent - I am convinced. Absolutely convinced - Will no longer record at any rate higher than 48K. Thanks. Further to recording, there is nevertheless a benefit for non linear plugins, and poorly designed eq's with cramping, to process at higher sample rates, and/or ensure that these plugins have been suitable oversampled internally, and if we use eq's that these eq's do not cramp at nyquist.(the upper limits of 1/2 the sample rate).

[@AWidgetIHaveNot]

Five thumbs up from me. I live in London. Can I be your tea boy? I'm not going to tell you my hearing is shot from too many Zappa plays Zappa gigs because who needs hearing when you have a DAW? And thank you Alan Parsons for my tinnitus.

[@iainparamor1482]

I genuinely can't hear a thing above 16.5k haha

[@tommibjork]

AMEN!!! Thanks for this, this is what I've known and promoted for ages. 24/48 since 2004. Still holds true today.

[@PresentDayProduction]

Thanks Tommi, and thanks for watching 👍

[@tommibjork]

@@PresentDayProduction it's odd there are many known producers telling people to "back up your material in highest possible frequency", in this context that makes no sense...

[@PresentDayProduction]

Tommi Björk Everyone knows 192khz stores better! Only if you keep your hard drives in the fridge though 😂

[@dweezz]

@@PresentDayProduction and use the correct cables

[@juhapeltola8232]

This is amazing! Great video to show what sample rates really means. I had't no clue :D

[@flashbak01]

Excellent presentation guys. Thank you.

[@chriscutress1702]

I agree with the general use of 24-48 for the benefits you mentioned. I can hear the difference between 16-44.1 and 24-48 with certain types of acoustic instrument recording but not generally with pop recordings. The only time I have ever heard the difference between 24-48 and 24-96 was during the recording and playback of classical music in a state of the art recording studio through B&W speakers. Generally just a little extra space around the instruments. Very subtle but was audible in a blind test. Other wise 24-48 is just fine for most pop-folk-jazz recordings. And anything that is over compressed and has no dynamic range then you might as well listen at 16-44.1 or (ARGHHH) mp3 as it generally has no air around the recordings to benefit from higher sampling rates. *** note that the classical 24-48 vs 24-96 listening test was about 12 years ago and I'm not sure I could hear the difference now that I'm 65 but I can still hear the difference between 16-44.1 and 24-48 on most acoustic instrument recordings via A/B comparison. I also notice that after listening for extended periods at 16-44.1 I can feel quite tired whereas when listening for a similar amount of time to analog tape playback I don't feel tired. Digital ear fatigue syndrome ?

[@beatweezl]

I busted out laughing at least three times in this video. You Brits are hilarious. Love ya.

[@myyt4382]

Hi there, I am sorry if I am a mental snail here, but could you please elaborate a bit on how did you test the converters? I do understand you have some test tone there but was there some loop estabilished or..what did you did exactly there please? I am really really curious to find out and test for my grandpa converter here :) thanks a bunch in advance for your time!

[@PresentDayProduction]

Use the test tones on our website - link is in the description. That will tell you if your converters are working well at higher sample rates. If you can’t hear anything but can see it on the meters then you’re good 👍

[@johnshotwell3803]

Very nice, great explanation. I am a theatrical sound designer (I record and produce sound effects and voice over bits (often distressed), and edit music, for live theatrical productions.) I started out working in (recording and editing) 1/4" half track, then I switched to minidisk. When I was first working with a DAW, I was rendering my sounds on CD, so I was using 16 bit, 44.1 for my recording. Currently I am rendering sounds to AIFF or MP3 files, I still work in 16 bit 44.1, I don't believe the differences I hear between that and higher resolutions, in my monitors, are noticeable in theatrical playback systems.

[@PresentDayProduction]

Thanks John. 1/4” tape... minidisc... then there was that weird transition period in the late 90s/early 2000s where nobody was quite sure how to get audio from one location to another! Thanks for watching 👍

[@benpyke9298]

As the proud recipient of a Music Physics degree, I love this question. You are completely correct..... for 95% of the video. The ultra-sonics CAN and DO interact with each other to create audible waves while beating against each other. This is where I must put forth that I live in "Live World" (part of why I just got around to watching this video) where 24/96 is kind of standard (most of our DACs to get to CAT5 are for 24/96 only) but we do use that ability to "hear" the ultrasonics from the mics, let them bounce off each other and create audible tones in the console, and bounce them out to speakers that only give us up to 22,000 Hz. The only thing I have a problem with is that you have proved for recorded sound being mastered 24/48 makes the most sense. I agree with almost everything, except that 96 doesn't need to exist, cause it does. Instruments create these sounds, we need to reinforce the WHOLE sound of the instrument, not just what WE hear from it cause that instrument and another will beat together and create beautiful harmonics that get lost with 44.1 or 48. (Again this is all in Live World).

[@PresentDayProduction]

Thanks for your comment Ben. Firstly, I hope the current situation we’re in ends soon and gets you off KZbin and back into live world! You guys and girls have had it tough this year, and we feel your pain. I do occasionally master at 96 if the client requests it, and some of my clients swear that 96 sounds better, and that may well be because of the points you raise above. And that’s fine, if it sounds better it is better. The main reason most of my work in mastering is at 44.1 or 48 is because I’d rather handle the conversion from anything recorded and mixed at higher sample rates here, so as I know what it’s going to sound like, and don’t leave it to the streaming services! I’ve done a fair bit of work in low-mid level live sound but you’re clearly in high-end world. Can you explain to me the reason for wanting to hear the ultrasonics from the mics and create audible tones in the console please? That’s a new concept to me, and I find it fascinating! Thanks again for your excellent comment, and I hope things can return to at least a little bit of normality again for you soon! Cheers, Ben! Mark

[@EpithetMusicTV]

i too am interested in learning more on this concept 🤔

[@Chris6741]

I'm a saxophone player. Thinking about how manipulating the overtone series can affect the timbre of the instrument makes me agree that there is inaudible information there but it might be information that is interacting with the audible range's harmonics. Of course that isn't taking into account the science of digital audio. It's just the way that I've pictured a higher resolution giving me a clearer picture.

[@benpyke9298]

@@PresentDayProduction Ahoy there Mark! First, thank you for the well-wishes -- being based out of the good ol' U.S. of A. I have a few more decades before this ends, at least the way things are going. I think you are right. I think for Mastering purposes especially, the difference between 44.1, 48, and 96 are.... negligible at best. I think that pretty much for everything you are doing, the difference is negligible. I would have been interested to see you record the same thing at the different frequencies to see if more changed or not. Frankly, I see no justification for above 96k. If we give some extra on either side for sensations but not "hearing" sounds we can say that humans are good for 10-22k Hz (and there is some interesting evidence that humans can sense frequencies higher than they can hear). Even so 48k is more than enough for this range of human sensing. So why 96k? Y'all correctly pointed out, mainly marketing. In Live World we also argue that having those ultrasonics in our console gives us the options. We have the ability, if your guitar makes a funky harmonic at 27k and another funky harmonic at 31k, with 96k I can hear both and I can hear the beating of them together at 4k which is audible. If I had only 48k that beating might be lost in the DAC. The other argument thrown around in Live World (which I am not a hundred percent convinced of) is that IF shit goes wrong for a second while computing in the console (lag, or error in memory or what have you) (I have had consoles entirely seize up on me mid show), then having those extra samples the compute can still recreate the waves that are in audible range if a sample or two here or there are dropped. Again, for what you are doing, I completely agree -- save the processing power. A lot of consoles will downsample for FX making the whole argument moot anyhow (not all, but many). I think that the only reason I would move from 24/96 is if someone comes out with a good peer-reviewed study showing that humans can sense more than the current max known frequency which (according to Physics and Music by White) is about 23k. That my reasoning. I greatly enjoy talking about this stuff though, and I would love a video about how you guys in recording world deal with this whole stay at home business. How has recording changed, was it already changed, does it piss you off as much as it does me to not have live shows? I would be interested. Thanks, Pyke

[@frankstetka7206]

Thank you Ben Pike; I was thinking of this kind of thing because I have noticed the difference in (air) if you will and this is what had me chasing high bit rates is cheap consumer recorders. Maybe I was hearing things but it’s true about how lower harmonics are affected. Interesting to note that people are often looking for flavor when drinking alcohol for instance. The same can be said for harmonic distortion and compression in a playback system. Maybe I do mind the information phase warping as much as the Perceived loss of dynamics which likely could be attributed to a lower noise floor or less perception of limiting on the high frequencies; all I know is that I have heard examples where 96k 24bit 44was far superior to CD @ 44.1 16 bit So at the end of the day it’s bit rate we are looking for. This reminds me of marking hype over watts in audio instead of efficiency and bandwidth. Cool stuff, great channel; great comment. 😉

[@Schwermetall]

Lol, please use a different kind of signal the next time for the sampling explanation. Try it with a square wave instead a sine wave a do it at a higher frequenz (more than 10kHz). With a SF44 every kind of 14kHz will be a sine wave. That's a part of the Fourier-Analyse! But don't get me wrong. To hear it is a different thing. Best regards, Alex

[@benjaminjoeBF3]

yeah too few points at 12k to reproduce anything than a sine. This can reduce hi freq details, def in the hearing range. Oh well digital sucks anyway ;)

[@benjaminjoeBF3]

@MorbidManMusic Remains Cant stand people having different opinions? Whats there that make your bum burn lol and best regards

[@TheJonHolstein]

square wave, is the same as a sine wave with sine wave overtones, so when hitting the frequency limit the shape will change, as the overtones are removed. Our ears do the same thing, so as long as the frequency represented passes our hearing, there should be no difference. But without anti aliasing filters, some issues can occur.

[@Schwermetall]

@@TheJonHolstein "there should be no difference"... well, that's the question

[@TheJonHolstein]

@@Schwermetall if all you hear is the pure fundamental frequency of a square wave because it sits at the upper limit of your hearing, that is a sine, no matter what it would look like on an oscilloscope capable of showing frequencies above your hearing, but filtering perfectly to match your hearing, the waveform on the oscilloscope would be identical to what you hear. There is no difference if it is correctly handled, but that requires filtering down (anti aliasing filter), to avoid the wave foldback, that occur when passing the nyquist limit. Our hearing though isn't like a brick-wall filter, and losses of frequency hearing, doesn't necessarily happens in a precise order, so it would take a lot of experimentation and loud playback, to find the aboslut top frequncy of ones hearing.

[@Bluescobra]

Hi, terrific video, many thanks! My conclusion: 24 bits for sure, 48khz is good for most people/applications, but when you have "know/trusted/great gear" 96khz is worth playing with, especially when using digital plugins behind.

[@weschilton]

See his point is that no is isn't.

[@stephenkennedy7347]

Good info and I will definitely try 48K. I converted to 96K a few years ago and I haven't had anything weird that I can hear but I will try a project at both and compare. Cheers!

[@recordingkc]

Endgame level explanation. Well done!

[@PresentDayProduction]

Thanks RecordingKC!

[@RondellKB]

Great production and very informative

[@PresentDayProduction]

Cheers! :)

[@MikeSadlerAU]

Great video; thanks Mark! I went back to 48/24 after several years at 96K... basically to get more simultaneous channels recording 'live'. There is no difference in quality using good plugins. The exception to 'bigger isn't better' though, is using e.g. Amp Modeling/IR's in a live or live-in-the-studio context... assuming you have enough grunt, higher sample rates will deliver reduced lag to the performer's cue mix and so for recording direct through e.g. BIAS Amp, I have a seperate machine running at 96KHz running the BIAS amp and Torpedo IR sims which *just* lowers the latency enough to make it indistinguishable from IRL with an amp two metres away, and gets that audio back onto t he main machine's interface in almost-real-time to line up with all the other instruments being recorded via 'normal' AD conversion on the main machine at 48/24. If it's not being recorded live, I obviously *just* record those parts at 96 and import them into the other machine directly with a down conversion to 48. Obviously, I reckon 96 is the go for effects/interfaces that are used live, for similar reasons, but I always keep in mind that 'sound' has 'latency' too... if you are used to playing e.g. 5 metres from your guitar amp on stage, the delay from a 48/24 wireless signal from your guitar to amp and then back to your ears via a digital IEM system is not going to faze you :-)

[@codyrap95]

Why didn't you cover the only use of high sample rate audio: Slow Motion or the ability to stretch it without much artifacts?

[@ezrashanti]

This is a relevant point. Also oversampling is important if working at lower sample rates to prevent aliasing. Also, higher sample rates give lower latency. I find 96k to be the sweet spot for audio production where all of the plugins work and the latency is good but the CPU load is not ridiculous and the file sizes are manageable.

[@codyrap95]

@@ezrashanti Well technically this would be impossible, because 96k needs more processing power by definition, so how would the latency be lower if we need to process bigger, higher "resolution" files?

[@andreatomassini5521]

@@codyrap95 I don't remember exactly why, but yes, working at higher sample rates gives you less latency(but not less cpu load of course)

[@codyrap95]

@@andreatomassini5521 Uhm ok, not gonna say no but from a theoretical standpoint it doesn't make sense to me. I will try that! Anybody has an explanation on that?

[@ezrashanti]

@@codyrap95 Technically you are wrong and need to do more research.

[@juap]

192khz is better you don’t know nothing, more is always better! ... I was joking, I’m currently recording at 44khz because that’s default on the Apollo Twin, and I read that if I record at 48 then export my mix to, for example Spotify, they downgrade to 44 and that process can change a little the sound... don’t know, that’s true? Thanks a lot!

[@PresentDayProduction]

Thanks for your comment! Yes, I generally send masters back at whatever the ‘destination’ frequency is, so I know what it sounds like, and aren’t reliant on any conversion their end

[@Crossfire2003]

Hello! Great video. Though I agree with you that 24-bit 48kHz is “the way to go”. I can objectively say that I have a 24-bit 48kHz album that sounds great. I’ve re-exported the album in 16-bit 44.1kHz and I was able to hear the differences in trebles.

[@PresentDayProduction]

That might be down to the sample rate conversion. Not everything does it particularly well! What software are you using?

[@Crossfire2003]

@@PresentDayProduction, I used Sony Sound Forge Pro 13 to import, and re-export the tracks at lower resolution.

[@BedroomGuitarHero]

Man, your channel is fantastic, thank you for doing what you do, you certainly deserve way more subscribers than you have! Please keep going :)

[@PresentDayProduction]

Thanks for the kind words!

[@MrMeluckycharms]

Magically delicious fun facts found in this video!