Lmao. My tinitua is a little higher.

Never been able to figure out where mine is as it doesn't seem to be a single tone.

7.3K in my right ear 😢

Hello 👋 👋 👋 I need a course of Signal Processing Can you help me Thanks in advance

@@yaakoubberrgio5271 I've put up the lectures from my ECE3084: Signals and Systems course at Georgia Tech: kzfaq.info/get/bejne/jNqDn9CV2M7Von0.html

A well designed delta modulation ADC does part of filtering for you. A part like the AD7768 uses a modesty high order integrator so you get more than one pole of anti-alias for free.

@@kensmith5694 I've never been able to fully wrap my head around delta-sigma converters. Like... I can sort of follow the math line by line, but I can't really develop and intuition for the "heart" of how they work.

@@kensmith5694 When I mentioned working in sonar engineering I was talking about 1979 and 1980. We had to construct our own processing unit by using a 5 MHz 32 bit 'high speed' multiplier as the heart of our system.

Wave scientist, here (not audio). I agree this is an excellent demonstration of aliasing. However, I think this video seems like an argument primarily for *mastering* above 44.1 kHz, particularly if you're generating a lot of synthetic sounds, rather than recording or playing back audio above 44.1 kHz. I wouldn't expect human voices or musical instruments to produce a lot of power in frequencies above the human range of hearing, so you're probably not going to get a lot of audible aliasing if you record audio at 44.1 kHz. And then if that aliasing hasn't been baked into your digital audio file to begin with, then you won't be hearing it. An exception I could imagine would be if you're recording in a noisy environment where the "noise" isn't Gaussian--in which case, perhaps you could get some beat-like pattern of "noise" in your audible range. Edit: the other caveat would be that if you have high-fidelity audio and you're playing at back at a lower sampling rate, it's anyone's guess how that downsampling/resampling algorithm is working. It might introduce it's own wonkiness. But then if you're trying to drive speakers at a frequency beyond which they've been tested, you might get non-linear weirdness, too.

I'm really focusing on the capture side of things. My interest is really in the "making" side of things. Some cymbals can shimmer up in the high range of human hearing - I've heard tell of some loss of the "brightness" of cymbals because recording at 44.1kHz and compounding of a bunch of low pass filters causes that upper range to just lose power... but that's theoretical to me. But I really never answer the question in the title "What is the Optimal Sampling Rate"... haha. I think my purpose was really to understand what the argument is, not necessarily advocate for it. That's my impression when I was trying to work through Lavry's paper.

@@FilmmakerIQ this is an interesting discussion and has me sitting here on a Saturday afternoon tinkering with MATLAB and Audacity.... :-) So what I just tried was I created a 7 kHz square wave sampled at 44.1 kHz. It looks like a typical square wave. Sounds like your video Then I generated a 7 kHz square wave in a 192 kHz track, and applied a bunch of aggressive 22.05 kHz lowpass filters so it has no frequency content above 22.05 kHz. Then I made another 7 kHz square wave in a 192 kHz track, and just told Audacity to resample it to 44.1 kHz The two 7 kHz square waves generated in 192 kHz tracks, one lowpass filtered, and one resampled, sound like square waves, and sound almost.the same. The one 7 kHz square wave generated in the 44.1 kHz space has dozens of overtones and sounds totally different. I don't have an easy to way generate a square sweep, but this would be an interesting experiment for you to try on a square wave sweep and see what happens.

So here's the deal with 7 kHz square wave and any flavor of 48kHz (96kHz, 192kHz) - they will all sound the same! Remember in my video where I did the frequency analysis? There was the fundamental at 7kHz - that first harmonic at 21kHz and then aliases separated by 2kHz starting at 1kHz and going up. That works out because cycle of reflections caused by the 24khz Nyquist limit cycle around and around on odd numbers. Doubling the sample rate to 96 or quadrupling it to 192 doesn't change the cycle, it just changes where the cycle ends and picks up again! And since a square wave is suppose to be an infinite series, it doesn't matter where the cycle picks up.... Change the fundamental frequency of that square wave to 7001hz - and it won't cycle around like that and you'll hear the difference between 48 and 192khz.

​@@FilmmakerIQ "aliases separated by 2kHz starting at 1kHz and going up. That works out because cycle of reflections caused by the 24khz Nyquist limit cycle around and around on odd numbers" - sorry, don't understand this: where does the 2kHz separation come from ? How do you get from 7x5, 7x7, 7x9, ... to all those frequencies ?

don't underestimate the degree to which lossy compression might actually be doing a better job of preserving the signal than you think - eg, check out dan worrall "wtf is dither"; it's a long video and I don't remember exactly where in the video he does it, but somewhere in the middle he compares mp3 to 16bit wav in a situation where the mp3 *unequivocally destroys wav* in terms of which one represented the data better. wav was more lossy than mp3. That's because quantizing to 16bit integer naively actually introduces more noise than mp3 compression, if your signal is simple enough. it's all about what bitrate mp3 or ogg needs in order to near-losslessly compress a given section; and ogg vorbis is based on wavelets, not discrete cosine, which was why ogg vorbis can handle certain kinds of phasing sounds much better than mp3. so - yeah, as long as you're in a high enough quality mode that the bitrate compression is in the -100db range, you'll probably be able to hear whatever -70db effect they're trying to show. it's only when you turn down to 240p and your mp3 noise is -10db that we have a serious problem from audio compression. now, video on the other hand... :D

In my experience watching a movie/TV show on Netflix and watching it on Bluray is usually night/day difference. Its not so much that you obviously lose highs, you seem to lose dynamic range, it sounds flat and dull. Of course its not always enough to spoil the experience, but sometimes it definitely is. Same with the picture quality.

disagree comoletely. YT audio is highly compressed and I can tell the difference between songs in YT vs Apple Music. No contest

@@jhoughjr1 Music videos strangely are often the worst offenders, whereas some youtubers use music and it sounds fine. I'm very sensitive to lossy codecs too. Hated Bluetooth audio until LDAC and Samsung Scalable came along.

@@jhoughjr1 i dont know super exactly what youre taking about. but ive heard youtube uses aac codec. Imho for certain bassheavy generes, youtube is miserable. bass just doesent translate well on it. guitars are ok, but i still preffere mp3s of mine.. Apple music also uses aac i heard. But i found it bit better, dont know if its a specialized aac version they use. Other than that i seen a test video that compared waveforms to lok for normal compression (audioplugin compression) , and nothing was found.

I mean, even cheap equipment like 100$ DVD Players in 2007 had already 192kHz DACs avoiding any problems like this at all. But for the final media, more than 44,1kHz doesnt make much sense since anyways most released music is still in 44,1/16Bit. Even most(or all!) vinyl records are made from 44,1kHz samples. Tidal even dare to upsample/"remaster" 44,1kHz/16Bit originals to expand their "HiRes" collection... Since anyways every HiFi gear have HPF for anything above 20kHz, in combination with internal 96kHz+ processing, more like 384kHz nowadays, no. 44,1 is just fine... more is acceptable in cases of digitalized vinyl records or yeah - why not. 44,1/48 vs 96+ is like comparing 4K vs 8K... it doesnt make practical sense, probably a bit with the perfect circumstances... but hey its possible. Thats why my AVR has 9x(or 11x idk) 384kHz/32Bit (32Bit!!! wtf?!) DACs, by numbers even better than my Hi End Stereo Gear with "only" 192kHz/24Bit Wolfson DACs. Only in recording and mastering more than 44kHz are needed, and these are anyways at 96kHz+ since its possible. I dont get people when they complain about "only CD quality"/44,1kHz... damn! Thats at least completely uncompressed, not like the lossy! MQA garbage for example. In fact (and already proven...) CD quality is better and more accurate than MQA (which is another compression format like mp3 - but worse, and with high license fees haha). Some of my friends are completely addicted to HiRes and/or Tidal/MQA, only because they see any blue light or 96/192kHz on their receivers screen... despite having absolutely the same sound as a 44,1kHz CD with the same mastering. Damn, they use soundbars, garbage "HiFi" gear, BT headphones and they dare to complain about 44,1 kHz only! I also prefer HiRes source material, but mostly because of different masterings, less loudness, more audiophile/dynamic, easily for the "demanding" people mastered.

@@harrison00xXx I believe you are incorrect about vinyl masters. Mastering for vinyl is a separate master than the master for CD. Professional mastering engineers want to work with the highest quality mix which means NOT 44.1kHz/16bit. And most likely the vinyl press wants to make their master from the highest quality version available. At least for the major label artists. Independent artists, well ya know they get what they pay for and can’t be reasonably used to make statements about what’s used to make vinyl records.

​@@arsenicjones9125 Ofc its differently mastered for vinyl, but still, the samples used to make the "negative" are for probably 99,9% of the (non quadrophonic) records 44,1kHz/16Bit, as CD quality, that was my point As if CD Quality is "bad"....cmon, thats the most accurate and "lossless" quality standard we ever got. Ofc there is now "HiRes" but thats more of a voodoo/too much...

@@harrison00xXx no I’m afraid you’re again incorrect. In major studio albums they regularly record at high sample rates then down sample to 48khz 24bit to edit & mix. Some major studios do all their editing and mixing work in 96khz/32bit floating. Then it will be down sampled again after mastering. Again we can dismiss what independents do because they don’t do anything in any standardized format. CD quality is not the most accurate, lossless standard available. 🤦‍♂️🤣 An original recording made in a 96khz/32bit wav file is more a more accurate representation of the analog signal. If there are more samples w greater bit depth it MUST be more accurate than a lower sample rate and bit depth. Just because you cannot discern a difference in every piece of music you hear doesn’t mean there is no difference or that there is no difference which affects the experience. Just to be clear I don’t think CD quality is bad just that it’s not without flaws either. Upsampling won’t increase fidelity in anyway but a higher sampled recording is higher fidelity

@@arsenicjones9125 So you have proof that the source material for making vinyls ins more than 44,1kHz? Sure they edit and master at higher bitrates, but the end result is mostly 44,1/16Bit sampled. This change probably with HiRes for the customers slowly, but its a known fact that 44,1kHz were used for vinyls FOR DECADES at least.

This is about extra noice introduced during processing of the audio. Not about the output format really.

That is really an insightful way to look at it.

@@FilmmakerIQ Hi John, You call me "insightful" again and I will sue! :-)

Need to put a low pass filter on that comment.

@@MyRackley Hmm, sadly i know mine doesn't at 65, but then i've played in too many bands with overloud guitarists, and in one case, a drummer who overhit his cymbals all the time, where we rehearsed in a small room. Still have a low level of tinnitus in my right ear, but luckily it's not really noticeable unless things are really quiet, and i guess i've become quite good (or at least my brain has!) at filtering it out of consciousness!

Yes, I haven't factored in pitch alterations.

That's not entirely true, actually. The Xiph video mentioned in the content here covers the waveform phase topic as well. The reconstruction filter post-DAC is basically turning discrete samples into a Bezier curve. Just sliding the sampled points around on the X/Y axis (if X is sample, and Y is word value -- i.e., the amplitude of an individual sample) will alter the resulting wave's phase. Another way to think of this is to imagine using a strobe light to capture an object moving in a circle. If the speed of the object rotating about the circumference was perfectly aligned with the flashing frequency such that there are exactly two flashes per revolution, it would look like the object is appearing in one spot, then another spot 180 degrees from the first, and repeating indefinitely. This is basically the Nyquist frequency. From that, you could construct a perfect circle because you have the diameter. So now, imagine altering the "phase" of that object so that the strobed captures place those objects at different places around that circumference. You can still construct a perfect circle. Same with audio samples. It doesn't matter if the phase changes. As the Xiph video says (I'm paraphrasing because it has been a while since I watched it), there is one and only one solution to the waveform created by a series of samples, _provided that the input waveform and output waveform have both been band-limited to below the Nyquist frequency._

@@nickwallette6201 Well, yes, for any arbitrary signal, you can still reconstruct it with sampling, but I was mostly thinking psychoacoustically, where delay and phase variations between ears plays such a big deal in stereo sound. And one of the side effects of sampling is that you get phase constraints, like I described above. For example, with a signal at half the Nyquist frequency, how do you distinguish between a full-amplitude sine wave and a cosine of -3dB intensity, when they both share the exact same sample representation (alternating between .707 and -.707)? Since that phase information can spell the difference between a centered (in-phase) or diffused (out-of-phase) stereo sound space, preserving phase and delay information is super important, and with finite sample intervals, there's only so many phase states you can have at high frequencies. I also acknowledge, however, that bandlimiting filters induce their own phase delays as well, which can have a significant effect on the perceived audio-hence one of the other advantages of higher sample rate is to relax the requirements of bandlimiting and reconstruction filters to minimize their coloration of the audio.

Delay is not an issue with the sample rate. Sample rate does not affect the precision of timing of the wave in any respect

@@eddievhfan1984 With two samples per cycle, you can reconstruct a waveform with any phase you want. You could indeed have phase and anti-phase waveforms at 20kHz with a 44kHz sample rate. Try it. Use an audio editor to create 20kHz sine, then invert the phase. Zoom in to the sample level and look at the waveform it draws. This is a representation of what the reconstruction filter does. I think it would be an academic exercise though, as 1) who's going to be able to determine relative phase between channels at the theoretical threshold of human hearing?, and 2) that's going to be in the knee of the low-pass filter curve, where any passive components on the output are going to affect the signal. It would not be unlikely to have a mismatch between L and R channels. High-end stuff might try to match capacitors to 1% or so, but there's plenty of gear out there (even respectable gear) that uses electrolytics rated at +/-20%. There's a lot of concern over perfection that is not at all practically relevant.

Yes but the how gets way more complicated

Yep

So that's why you can hear this high pitch noise from CRT TVs?

39 and oh gods do I NOT miss working on TVs and that wretched noise. I can only imagine how horrific that nose must be to cats and dogs. We practically used to torture our pets with those damnable things.

yep. as a kid I could hear if a TV was on even if the screen was dark.

I remember as a kid freaking want to smash all school tv’s what a trash they let us watch in the first place and then the fucking beep, will hear even now I think, I did run out the classroom sometimes and told the teacher to blast herself with this earpiercing beep! She was like: what beep!? Bitch.. the older the crt, the more chance you may use it to deflect vermin out of your garden..

CD's rule baby!

My earliest memories from the early 90's regarding CDs is that, a) they sounded really, really good, and b) my mom will get REALLY mad if we play with her discs (they were expensive)! My dad had a Panasonic component stereo setup, nothing high-end or audiophile grade but it was half-decent at least. He had some Type-II cassettes too which sounded really good on that player. By the mid to late 90's CDs were starting to replace cassettes as the on-the-go medium for portable players, boomboxes, and car audio, which tended to sound bad to start with, but no matter how good your system is all of these are frankly crappy listening environments. Whereas vinyl was never a portable medium so even now if you had a vinyl player you'd probably have it in a dedicated listening room at the very least.

1st Harmonic with 3 times fundamental frequency? Where is harmonic with 2 times frequency?

@@peteblazar5515 the components of a square wave are the sum of infinite _odd_ harmonics. So the first harmonic is 3x the fundamental frequency, the next is 5x, and then 7x, etc.

I wouldn't rag on mp3s either. Unless the bitrate is really low or it's encoded with an old encoder I just can't tell the difference.

Yeah I got that misconception as well... But there's another issue that shows up as low frequency "beating" when you start to get close to the Nyquest limit but still under. I don't know what it's called but it's worth looking into. The brick wall limiter was definitely the issue, I think it's particularly prominent in a mastering scenario... After you add up all the little tiny decrease in high frequency in all the audio stages plus all the filter chains... it becomes substantial enough to notice.

In the old days that was an issue. These days ADCs are sampling much faster internally and downsampling on the output, so the brick wall filter requirements are much less steep.

Fascinating analogy

@@FilmmakerIQ at the end of the day it's all about sampling at a limited frequency. Doesn't matter what the data is

Sorry but three sample points do not produce a sawtooth wave, it's produces a sine wave. You don't connect the dots with straight lines, you draw a sine wave through the dots. A saw tooth wave has integer harmonics, it would need to be constructed with many sine waves which works probably be above Nyquist if the wave is only 3 samples wide. Lastly, I don't think you understand why Lavry suggests 60. He stated in the paper that 44.1 is if not perfect, close to perfect.

@@FilmmakerIQ I think you misunderstood what I said - "triangle", not "sawtooth". And I wasn't referring to an actual triangle wave, I was only referring to the shape created by the three points if you connect them, which isn't exactly what's going to happen in the DAC anyway, because DACs don't transition from one point to the next in any smooth way, they simply jump to the next value. The bottom line is that for a 16kHz sine wave, only three data points are created, and only three data points are going to be output by a DAC. The DAC itself is not going to "draw a sine wave through the dots". It's just going to output stairsteps at three data points and that's it (unless of course we're talking about oversampling, which would instead use spline interpolation or some similar approach to approximate where the additional samples would be. But to my knowledge no production hardware - such as Pro Tools or UAD Apollo etc. - utilizes oversampling on output). For example, if you create a 16kHz 24-bit sine wave at -3.0db, each cycle will have exactly three points - one at zero, one at -4.2 db above zero (sample value 5,143,049) and one at -4.2 db below zero (sample value -5,143,049). The DAC isn't going to transition smoothly between those points, it's simply going to output a zero for 20.83 milliseconds, followed by a sample value of 5,143,049 for 20.83 ms, and then a sample value of -5,143,049 for 20.83 ms. If DACs did indeed "draw a sine wave through the dots", then aliasing wouldn't be a problem, because the DAC itself would be reacting perfectly to the INTENTION of the data - just as analog tape used to do. But the problem is of course, as with many things computer-related, DACs simply don't do that. They just output a voltage corresponding to a number for a specified number of milliseconds as dictated by the sampling rate. It is of course this behavior that causes the alias frequencies to result, as you have very correctly and articulately described. As for Lavry's 60, correct me if I'm wrong, but my understanding is that the advantage here is twofold: 1) it pushes the vast majority of alias frequencies into the supersonic range, making them a non-problem, and 2) it provides more headroom for creating anti-aliasing filters, should a playback hardware developer choose to do so, which sadly, very few ever seem to. My point was merely to essentially agree with Lavry, but I'm suggesting that when taking into account the fact that digital hardware designers prefer to do things in powers of 2, that a better choice for "optimal sampling rate" should be 64kHz specifically. Personally, I wish hardware developers provided that option in addition to 48k and 96k because that's what I would use for production instead of 48k or 96k. It would be quite a good compromise.

That's completely incorrect. Yes the DAC does draw a sine wave because it's coverting it back to analog. The speakers cone is a physical object and it moves through space with inertia, it can just jump to each sample point and hold for the next one. So If you produced three samples you will not get a triangle, you will get a sine wave. Watch Monty's video in my description. Samples are not stair steps, they define the points of a sinusoidal wave. This is the key to Fourier transform and Nyquest theorem. Aliasing has nothing to do with stair steps (because there aren't any stairsteps). Aliasing is the result of frequencies that are higher than the sampling frequency. Your understanding of Lavry's 60 is incorrect as well. It doesn't push alias frequencies in the ultrasonic... You don't push alias frequencies... it provides enough headroom for anti aliasing filters to work without affecting the audible range. Lastly clock speed has zip to do with binary. 64khz is meaningless because time is irrelevant construct. Look at the history of computing you will not see any clock speed correlating with any binary numbers... Because it's been simply not how that works... Also 64khz isn't a binary number. The closest is 2^16 which is 65.636khz

Same, I can't tell the difference between those last 2 waves but I can hear the transition. Almost sounds like there's a short crossfade

There is a short cross fade. I couldn't get the waves to cut exactly at the same amplitude so it was either a crossfade or a pop on the switch and I chose the cross fade.

Analog is not "infinite sample", your tape has a frequency range based on how fast you run it, normal copper wires will struggle with RF frequency, even the air has a frequency range because it's made of individual molecules. Natural is more "digital" than "analog" in the sense that energy comes in discrete packets because of quantum mechanics.

It has to do with the hairs in the cochlea of the ear. The ones responsible for the highest frequencies are in the smallest part of the cochlea (they have to vibrate the fastest). As we age, the cochlea becomes more rigid and inflexible to those high frequencies, and that's why we lose the high range.

@@FilmmakerIQ oh my…

>> percussion sampled at 44 Khz sounds like so much white noise but sampled at 192 Khz sounds like percussion instruments...

Nyquist will accurately reproduce the sound, if you THEN add extra modifications on that then that in no way implies anything about nyquist not being 100% correct.

@@ABaumstumpf Nyquist is correct about the absolute minimum sampling rate, but there are benefits in oversampling.

@@TurboBaldur Yes, of course, but that in no way has any effect on what us humans can actually hear, and there the 44kHz 16 bit is enough. if the mastering of the audio is done poorly that is not the fault of the medium not does it make Nyquist any less correct.

@@ABaumstumpf exactly, if the sampling is being done for playback to a human only then 44.1k is fine. But if you plan to edit the audio it makes sense to get more samples, even if the final export is to 44.1k

This is a great point, and I believe it may be why many digital recordings made in the early 90s sound "flat" compared to late-generation analog recordings. Too many engineers just relied blindly on the digital technology without thinking of consequences like this. Nowadays of course studios work with much higher bitrates and bit-depths for processing and mastering before producing the 44.1kHz or 48kHz files for release.

You choose a special case which is square wave with the frequency of the sample rate divided by four! There's two ways to think about that. Either the mathematical sum as you described or as a visual graph. Only one sinusoidal wave can fit the given samples... Instead of the sample defining the top of the square wave, it defines each side of the crest and trough of a sine wave with greater amplitude!

I agree.

These concepts are connected however.

LOL!

Hey John, I’ve been doing digital signal processing since 1980, 41 years, including spatial digital signals. Nyquist can be grasped with knowing one concept: that sampling at the Nyquist frequency there is no phase information. Phase information is restored as the sample rate is increased above Nyquist. To differentiate a square wave from a sine wave, both still have to be faithfully reproduced, including the phase information. At 10 KHz, a 44.1kHz sample rate only produces 4 samples her sine wave, partially preserving the phase of the signal. Since a square wave is made up of more than one frequency, the phase information becomes important, as it affects the sound not just the amplitude of the sound. 44.1 kHz works because most of what we listen to is under 8kHz. If you want to preserve phase up to 15kHz, really should sample above 60kHz. Now, if you are listening to stereo, you really want to preserve more phase information, so makes even more sense to go 60kHz or higher. Even though to me 44.1 kHz seems fine enough for me. I always wanted to make a spatial audio standard that recorded phase information as well as sampling information, a transformation rather than sledgehammer sampling. This has been done commercially outside the audio industry for over 35 years.

You are totally ignoring the sound reproduction equipment's role in this. Sure at 10 KHz, a 44.1 kHz sample only produces 4 samples. So? The signal recreated by the DAC sent to the vibrating membrane or paper cone of your headphones or speakers while reconstructing this 4 sample pulse of 1 second, it's plenty. 60 kHz may be useful during mastering of the original, but at the consumer level, we don't benefit from it with proper noise cancelling and anti-aliasing applied.

I'm firmly in the increase bit depth camp as well - but from my perspective, it's just buying insurance. I shoot a lot of stuff where I can't really monitor the audio - I'm just capturing everything - and with 24 bits have a LOT more room for error in the volume.

@@FilmmakerIQ The nice thing is that relative processing and storage cost is constantly going down, and there's always that consumer who's willing to pay for "the best." Who knows, maybe in our lifetimes 480kHz 64 bit will become mainstream haha

As was said here, it's cheap insurance, so why not. But, TBH, even good 16-bit converters are already near, at, or better than the noise floor of the analog signal chains on either side. (Especially when you start digging into the technical details of dithering.) Even if you think about the absolute top-shelf reproduction chain some crazy audiophile may have, with a home mortgage poured into their Class A amplifiers and directional speaker cables held up by non-resonant cable guides.... The studio was still combining a dozen tracks together (combining each of their noise floors) through a sound board with a gazillion passive components and a ton of make-up gain after the summing stage, sourcing each of those channels from pre-amps, EQs, and compressors built in the 1960s for that warm analog sound.... Did ANY of that equipment have a -120dB noise floor? God no. When combined, do you think there's any chance that a good CD player's DAC is the bottleneck? :-) About the only thing 24-bit (or higher) DACs can do is handle those digitally-generated fade-outs with a little more accuracy. Again, in the recording chain, there is actually incentive to use higher bit depths: To provide margin for error. In most editing suites, all source material will be converted to 64-bit floating point values on-the-fly anyway, and only re-quantized to integer samples for playback or bouncing to the master files. But still...

@@nickwallette6201 I didn't even realize directional cables were a thing. How does that work?

@@shootinbruin3614 Your guess is as good as mine. Probably about as well as using hospital grade AC outlets, or coloring the edge of CDs with Sharpie to prevent light refraction.

Okay I've gotta do a video on this because that analogy is completely wrong. Also analog tape has less specs than 16bit 44.1.

@@FilmmakerIQ analog tape has way more dynamic range and headroom than 16-bit audio at 44.1KHz. That's why everyone was still recording to analog tape, well after the CD, DAT and other forms of digital audio were invented. Believe me, they didn't do it because it was easier or saved money. Maintaining an analog recording studio with massive tape reels was an expensive and fiddly endeavor, so anyone running a studio back in the day would've jumped on the latest technology if it would've simplified that process. It wasn't until everyone eventually converted to digital recordings in the 2000's, when sample rates and quality of studio gear were high enough to record 24-bit audio, at sample rates well above 44.1KHz. You don't have to like my analogy, because it's not exactly perfect, but people know more about editing photos and videos these days than they do about audio--and they just need something they can wrap their heads around, to know why people choose to record at higher sample rates than what we hear as the finished product. However, my explanation and analogy are not wrong--let alone completely wrong. I not only studied digital audio in college, I also worked in radio, and even helped teach a class on digital audio production. The professor wasn't the most skilled at recording and editing, because he came up in the analog era, and just used the computer like a tape deck and did everything old school. So, I helped him teach students one-on-one, how to actually use a DAW in one of the studios, so that they could record their assignments. I still record, mix and produce music for myself and others in my spare time, so I might not be a KZfaqr but I know what I'm talking about, and I'm not sure you know what I'm talking about, because if you did, you wouldn't call me "wrong" and use that as the catalyst for making a video to correct me. I have no idea what your credentials are or experience in this field is, but I got the impression that you're someone who has some technical understanding, and just learned all of this shit in the process of making your video, and you really don't have more than a decade of actual knowledge. It's funny, because this video was actually lacking some pretty basic information about the topic. You didn't even explain why someone would want to record anything at 44.1KHz, when there are much higher sample rates. You brought up using 48KHz as the sample rate, but didn't explain where that comes from. I think your viewers are even more ignorant than you on the subject, and might not know that CDs happen to use 16-bit @ 44.1KHz, and that DVD audio uses 48KHz. For anyone else reading this that actually cares to learn something, CD's compromised on the sound quality, because they couldn't make players that played back compressed audio without making them super expensive, and that was the highest quality sound they could use and still fit an entire symphony onto a single disc. (Audiophiles are historically fans of classical music, and when you're launching a new music format that's only going to be be affordable to the wealthy and/or those with "discerning taste", you kinda want to make sure you can cater to them a bit. It was a huge selling point for anyone sick of flipping albums to hear the second half of the performance, and I'm sure that without the support of those snooty weridos, CDs might never have taken off.) DVD's used 48KHz because it was the base sample rate used by DAT, which was one of the original digital recording formats, and because it was what people were using in studios, it got adopted by Mpeg2, DVD and digital broadcast formats. It only sounds slightly better, and it's almost imperceptible if someone uses proper dithering when creating the final audio file. It was simply a matter of compatibility with existing pro-audio equipment, which also supported higher sample rates like 96KHz. Good studios would record at the higher sample rate, and then downsample their work for the finished product. DVD-A used 24-bit audio @ 48KHz, because they were purely an audio experience, so they could use up more of the space on the disc for higher quality sound. Newer formats like BD (and the new dead HD) DVD used 96KHz, again, because of the larger amount of space available. Which is still really good sounding, but it's still only half the sample rate of the highest quality digital recordings, which is 24-bit @ 192KHz. There may eventually come a time when there's equipment that can capture audio at a higher sample rate, but even the obnoxious audiophile community that would typically support anything that's higher quality, just for the sake of it being measurably better (even if it wasn't perceptibly better) hasn't been pushing for anything higher. Turns out, even they can't tell the difference between 24-bit audio @ 192KHz, when compared to a super clean analog recording, from a well maintained deck with Dolby noise reduction. If you don't overdrive the tape, or have it distort in the upper frequencies, and you play it back on equipment that doesn't have any ground hum, it sounds fucking amazing--and so does 24-bit audio @ 192KHz, which I guarantee you've never heard in your life. Unless you're in a legit recording studio with high end gear to hear the difference, you can't tell. You can absolutely hear the difference between analog tape and the much lower quality audio used by CDs, because the dynamic range is reduced to 96 dB (which is a non-trivial 48 dB less than 24-bit audio) and more importantly, it's less than the 110 dB range of analog tape when recorded using a Dolby SR noise reduction system. 32-bit audio hasn't really taken off, because 24-bit audio already overkill with a wide dynamic range of 144 dB, which is already higher than the theoretical dynamic range of human hearing, which taps out at 140 dB--so 192 dB is just needlessly wasting storage space. That said, 16-bit audio with proper noise shaped dithering can have a perceived dynamic range of 120 dB, but again pure analog tape also has an effectively infinite sample rate, so that combined with the actually greater dynamic makes it sounds better than CD audio. Honestly, I'm not even sure what the point of your video even was, because KZfaq isn't the platform capable of even showing the subtle differences between audio using sample rates of 44.1KHz and 48KHz, especially when KZfaq already filters out everything over 15KHz. You may not be able to hear sounds over 15KHz, but I still can, and at this point if your hearing is already damaged enough to the point you can't even hear a sine wave between 15-20KHz, then you're clearly not the guy who should even care, because those sounds are for you, and I would agree that you shouldn't invest in anything better than CD audio, because it's completely lost on you. For those of us that actually understand digital audio, and have fully functional ears that can hear everything from 20Hz to 20KHz, there's plenty of reasons to record or listen to music that's using a higher sample rate and bit depth than CD audio. Of course, that's just a simplified explanation of some of the vast amounts of information your video was lacking, because I didn't even discuss the bit rate of digital audio (mostly because we were discussing uncompressed digital audio, and it's only when compressing audio files that bit rate becomes an issue, because that's where the sound quality gets drastically reduced.) But hey, you're just a guy who doesn't really have a background in this stuff, so I don't expect you to talk shop on the fine points of all this. Those of us who work with this stuff for real actually need to know about how our recording medium actually works, and we have to know how audio works, so that when we're mixing it for your consumption, that it sounds right--so we don't expect laypeople to know how the Fletcher-Munson curve affects our hearing during the process of recording and mixing, or on playback over a sound system of any kind. So, while they title of your video isn't wrong--the work you showed to get to the right answer is, because nobody in the history of the music and recording industry, or tangentially film and television, ever said 44.1KHz was optimal. The reason it's not optimal, is because the low pass filter is still attenuating frequencies within the audible range. So when Harry Nyquist figured all this shit out, he was merely pointing out the bare minimum that audio had to be sampled at to reproduce the full range of human hearing. He wasn't wrong, it's just that there's no perfect low pass filter that exists, capable of attenuating frequencies outside the range of human hearing, without attenuating audible signals. So, even with the best possible filter, you're still going to cut things off well above what we can hear, just to make sure nothing gets cut. In the real world, I typically don't allow my mixes to contain very much above 15KHz, because as you've noted, it's not supported by KZfaq, and most people won't hear that stuff anyway. However, I do allow reverb to contain as much high end content or "air" as we call it in the business, because those are the subtle things your ears will detect and miss if it's unnaturally chopped. It's like bad lighting in a poorly edited photo, or CGI--you have to be an expert to know what you're looking for to see it, but we instinctively know when those subtleties are lost and it will seem wrong or fake. Anyway, good luck with your channel. Hopefully you spend some time learning and doing some research before you go off and make something that's going to confuse or misinform your viewers.

I'm not reading this novel especially when you start with a completely false statement that tape has more dynamic range... There's no point when you're so off base from the start.

@@FilmmakerIQ Maybe if you read what I wrote, you'd actually learn something smart guy. Feel free to look it up. Analog tape recorded with Dolby SR noise reduction, which was the standard in professional studios, had a dynamic range of 110dB, while 16-bit digital audio has a dynamic range of 96dB. I'm not talking about cassette tapes here bud, I'm talking about 1/2 tape used in professional studios to make multi-track recordings. So, please just STOP with your nonsense, because you don't know what the hell you're even talking about. You looked some things up on Wikipedia, and think that you're a professional because you make KZfaq videos. How many professional studios have you been in that actually had 1/2 tape machines? I guarantee you've never even seen a 1/2 inch tape in your life, let alone heard one played back over the studio monitors in a real studio. Clearly, you seem to fancy yourself a "Filmmaker" and not a recording engineer, or producer--so why don't you go make your silly little videos about lenses, or light meters, because you don't know shit about digital audio or recording.

Of course it wasn't nefarious... But it was one annoying obstacle in trying to demonstrate this concept. And then it's only on SOME of the streams, not all...

I've heard that too -- but unless they have special scientific microphones designed to capture frequencies above human hearing, I'm not sure it matters.

I don't think it's about frequency width. It's about stretching the entire recording. When you stretch it makes everything thinner. Like pulling rubber band. Signal would get less resolution. Less data points. Through entire range of frequencies.

This guy talks about pitch and time stretching uses 96k recording. Sound effects for movies. kzfaq.info/get/bejne/adVdhshp1rfIdJ8.html

Reasons for 96k recording. Time and pitch stretching. kzfaq.info/get/bejne/adVdhshp1rfIdJ8.html

Problem is the rubber band analogy doesn't work because Nyquist does not work that way. Using 24khz, the audio would be EXACTLY the same as 48khz in every respect BUT only up to 12khz. So it's not that you have more data points - that doesn't matter when the audio is sent back to analog in the speakers. I suspect the reason 96hz would be used for slow downed effects is the same reason I discussed in the video: headroom. With 96kz there's about an octave and change you can manuever around in without running into a Nyquist limit that dips into the perceivable range.

AAC is used for Apple devices which is locked to 44.1. It also happens to be what they use for the download file option in YT's creator studio.

Aha. Interesting. Using youtube-dl, here are the streams available for your video (limited to audio): 249 webm audio only tiny 52k , webm_dash container, opus @ 52k (48000Hz), 7.13MiB 250 webm audio only tiny 61k , webm_dash container, opus @ 61k (48000Hz), 8.37MiB 251 webm audio only tiny 108k , webm_dash container, opus @108k (48000Hz), 14.81MiB 140 m4a audio only tiny 129k , m4a_dash container, mp4a.40.2@129k (44100Hz), 17.71MiB I'm on a mac here (but not an iOS device); in Firefox, the youtube web app uses stream #251 (as visible in the "stats for nerds" right-click; in Safari it uses #140, so you are indeed correct! Again, thanks for the excellent video.

Yeah I over stated the Gibbs part

Well, there are music styles that use aliasing as a stylistic element. But one thing dynamic distortion has going for it is that the frequency content it adds is harmonically related to the input and even with intermodulation (where there's a risk of losing this feature) you have some intervals where it remains so - hence the popularity of power chords, where the added frequency is an octave below the root note. So I doubt it will become something with popular appeal and your reasoning reminds me of the old "we went from triads to tertrachords and now romanticism is regularly using quintachords and sextachords, so obviously the next big thing will be 12-tone music".

I would add that you don't want those higher frequencies even if you could feel them. There's nothing musical about those frequencies

@@FilmmakerIQ While I would agree with that statement, that is based on taste/emotion and as such can be argued against by those of different opinion. The rest however, is based on science/engineering.

No it isn't based on taste or emotion, those frequencies are quite painful to listen to. I had someone else comment that the experience trying to hear a 21khz tone was like shoving needles into your ear.

@@FilmmakerIQ :-) I find valve amplifiers sound woeful, yet a number of people with deep pockets swear by them. This is that same sort of thing. Agreeing with me that it brings nothing to the occasion does not stop someone somewhere thinking it sounds great, just like I like birds eye chillies and my wife doesn't.

Not all MP3 encoders cut at 16KHz though . LAME Encoder does not beyond a certain bitrate. And anyway, KZfaq does not use MP3 compression. It uses either AAC or Opus.

Yeah, I dyslexia

we don't need to hear anything about 10khz ( except harmonics)

You can hear the difference but did you hear the fake out I pulled ;)