D/A and A/D | Digital Show and Tell (Monty Montgomery @ xiph.org)

[@vedritmathias9193]

The showmanship in this video is astounding

[@soloperformer5598]

But misleading.

[@jacoby6000]

​@@soloperformer5598how is it misleading?

[@OMNI_INFINITY]

And distracting

[@jessegrisham]

@@SamsungTshirt Its not, he just had to come in here and make sure you knew.

[@eriktomas9194]

Not a chance, it's styled like a tutorial for elementary schools. I don't see how that's a problem.

[@caremeprenant]

I've been a sound engineer for a long time now and I've never watched anything as clear and perfect about digital audio! Thanks a lot!

[@SoloStuff]

I've never seen such a technical video explained in such a great way. Thank you, saved me a couple of readings.

[@Jon_Music-uv4vb]

This is, by far, the best demonstration and explanation of this subject that I have ever seen. A true mic-drop moment!

[@quantuminfinity4260]

This is so well presented it’s crazy!

[@6sixtysix]

In 100 years this video will be a treasure

[@VictorVonBelmont]

It already is.

[@ghreghdehomeshrhamesh8639]

It always will be...

[@lewiswaddo5045]

Still is! 😂

[@JRob1125]

11 years later and this debate is still being had....when this video came out I thought it was the death blow to the "staircase" argument. Man was I wrong. Lol For some reason, people feel the need to justify listening to technically inferior formats. I love listening to vinyl just as much as CD or streaming despite its flaws. I wish more people could just admit that

[@robertendert8944]

This definitively is the best video on this topic I have ever seen. Very instructive with excellent experimental set up and great visual overlays. It's hard to believe that even after this video people keep on spreading the well-known myths in digital audio. I would love to see more of this video's by Monty!

[@guyboisvert66]

I am an electrical engineer myself and this presentation is really top notch! We hear/read so much plain false stuff about audio and the famous "Analog vs Digital" debate... Cool if people prefer analog gear, it's "colored" a certain way they like. This was hilarious to see the reaction of some "analog guys" on youtube saying "this recording deserves analog" and always talking about the "vast superiority" of MSFL "all analog" recordings when suddenly they learned that MSFL was recording majority of their releases using DSD (which is a wise choice) ! Not to mention people paying thousands of dollars to get reel tape format music! The audio / music market is highly "modulated" (!) with mercantile goals. The other concept that average joe doesn't understand is that the transfer function of each component, listening room, the ears + personal choices and finally "placebo effect" are the heart of this endless debate! I'm not debunking analog, it's a personal choice like any other. All i can say is that i'm not missing a single second the analog sources i had before! (Nakamichi BX-300, Linn-Sondek LP12, Rega Planar 3, etc)

[@artysanmobile]

Monty, the perfection of your explanation is, once again, completely lost on a group of people who insist on believing their preconceptions. What a shame. At any rate, thank you. Perhaps one hundredth will be inspired to pursue the brand new understanding they will need to finally hear the penny drop, rendered in perfect analog sound.

[@hi-fi-guru]

What a GREAT explanation that is understandable about a misunderstood concept. Thank you.

[@davecool42]

I show this to analogue purists and audiophiles on a regular basis.

[@WindsurfingNelson]

Impressive! Excellent presentation. Thank you!

[@soloperformer5598]

Another one conned.

[@derpz_]

what is your problem?@@soloperformer5598

[@cstefanile]

Still the GOAT vid on the subject, hands down.

[@user-uw5xc2ms2j]

I was shouting like it was a sports match, and was team is winning, but I wasn't watching baseball -- what I was looking at was an interpolation plot. And then my students saw me yelling and shaking my fist joyfully at a graph

[@Mtaalas]

Every single DAW maker out there should just reference or mirror this video series... maybe people would start to listen to the engineers behind software they use daily who have to KNOW this as fact and stop yellinga bout vinyl being "better" or digital sounding "harsh"... digital sounds exactly what you put into it. no more, no less. So it sounds harsh if your signal is "harsh"... but i'm against using these emotional words to describe some phenomena that we can measure. Though, that's what many people do... maybe that explains a lot?

[@stephenyoud6125]

Wow - great demos and explanations

[@CisterJr]

what a perfect video!

[@soloperformer5598]

Kinda!

[@tmjcbs]

It's too much effort, but I would like to put a link to this video under every video where they advocate high-res audio and/or mention the continuous analog signal vs the stairstep digital signal or other such nonsense. So few people know about the Nyquist theorem...

[@lindsayandrews5707]

Just came across this video randomly. As someone who has worked for DECADES with digital audio, I found it amusing, refreshing, and very informative! Well done, man!

[@FunkyELF]

This is the best video on the internet

[@harold2718]

That shaped dither felt a bit like the "constant background noise" (that thing that goes away sometimes after swimming). IDK if I like that, but of course it's normally much lower in volume

[@thewhitedragon4184]

It is. Dithering is technique for preserving detail when reducing bit depth. Think of a bit crusher, it gets sharp noise artifacts when you reduce bit depth. The classing bit crusher just introduces aliasing artifacts to your sound. Dithering on the other hand is adding white noise then bit crushing. This white noise randomly increases or decreases the amplitude of your signal so when it rounded to the nearest small bit depth value, it might be rounded to a bigger or smaller number then it would have without the noise. So the dither IS just noise.

[@goldenretrogames]

This video deserves to have MILLIONS of views. There's still so much misinformation going around about digital audio.

[@ParboiL]

Such an interesting video and only 17,000 likes. Thank you very much! 👍

[@travails3829]

Interesting and I learned quite a bit. The outstanding question I have is whether this apparent signal integrity is maintained with real-world recordings, rather than these examples which use a single frequency.

[@FM-kl7oc]

There is no "apparent integrity". Below the Nyquist limit there is only total integrity of any signal -- no matter what the content of that signal is. It doesn't matter if the signal is a pure sine wave, which in a Fourier transform would be the least information dense signal possible, or white noise, which would be the most information dense signal possible (same as picking a true random value for every sample). This is all covered by the Nyquist-Shannon sampling theorem.

[@srpenguinbr]

It's a linear system. Since all band limited signals can be separated into a spectrum of frequencies, you can analyse what happens to each frequency to understand what happens to the entire signal.

[@jessegrisham]

@@srpenguinbr Excellent and concise explanation. Bravo.

[@dlarge6502]

All "real world" sounds are nothing more than single frequencies added together. In fact Monty already showed a square wave, which is infinite frequencies added together. But as we are all band limited (our ears) we can only hear the first 19 or so, and even then you'll have to be a child. You'll find it very inconvenient to demonstrate these examples if you were not using a single frequency, you might be able to handle a few sine waves but what is the point? Monty already showed you 19 sines added together.

[@travails3829]

@@FM-kl7oc Exactly, and since we don’t have infinite data bandwidth, the integrity would surely suffer.

[@AvithOrtega]

An oldie but goldie

[@SomeoneOnlyWeKnow.]

This video is so good and useful

[@jamessun7851]

He is using a oversampling DAC for the demo If he uses a NOS DAC, the result wold be different. You will see stair step signal output

[@user-yk6uy7su9i]

It's an analogue reconstruction filter after the DAC, whether it is OS or NOS does not matter here. If it was without reconstruction filter, the signal output would indeed look different, but the ear would do the analogue filtering so the acoustic result would be the same.

[@jamessun7851]

@@nicksterjNot sure if KZbin remove my comments or it was removed manually. Let me do it one more time here. Please search for the page "how-to-pick-the-best-filter-setting-for-your-dac". On that page, it shows a real stair step waveform output from a modern DAC, Topping E30, using NOS mode (implemented by using zero-order hold as you mentioned). The output for a perfect 1kHz digitized sine wave still look like a sine wave with many steps. For the perfect 10kHz digitized sine wave, you don't want to see its audio output. The very, IMHO, simply misleading by suggesting that you won't get the stair step output. 🤨

[@jamessun7851]

@@user-yk6uy7su9i Not sure if KZbin remove my comments or it was removed manually. Let me do it one more time here. Please search for the page "how-to-pick-the-best-filter-setting-for-your-dac". On that page, it shows a real stair step waveform output from a modern DAC, Topping E30, using NOS mode. Topping E30 does have analogue reconstruction filter but it is not good enough for NOS mode. The output for a perfect 1kHz digitized sine wave still look like a sine wave with many steps. For the perfect 10kHz digitized sine wave, you don't want to see its audio output. Yes, I agreed that you would still be able to somehow filter the stair step waveform with your ear but the point is that this video is misleading as you could still get stair step waveform from a even modern NOS DAC (or modern DAC with NOS mode)

[@slofty]

No you wouldn't, and that's why you aren't an engineer with a degree who has suffered through linear alg like the rest of us did.

[@user-hn8fw6eb2s]

This is wonderful. Thank You. It's amazing how a device creates the function that goes through all the dots in real time. We need more like you. There are people selling USB "reclockers", and "acoustic dots" out there. Is there a reason to seek an analog oscilloscope over a digital one ? If not, do you have an inexpensive old digital one that you would recommend ?

[@SpencerTwiddy]

TheWhiteDragon, whoever you are, you are one of my new favorite people.

[@chillyvanilly6352]

excellent video, thank you very much! But, it might have been nice to include the answer to the question "What is quantization?" :)

[@koimaxx]

9:38 It's the conversion of a continuous value (analog) into a discrete value (digital). You can imagine it as the rounding of a decimal number (3.14159) to the closest integer (3). Hope this helps.

[@d3d_compiller47dll]

This explanation is perfect

[@soloperformer5598]

But limited.

[@honeywellparts7541]

@@soloperformer5598 "BuT LiMiTeD" Bet you are one of those people that use the "Your audio isn't distorted enough" argument. 🫵🏻🤡🤡🤡

[@whoiscm]

I can pass my class. Thanks god for this video.

[@scottfaircloff9530]

thanks for this!

[@kenclarke5966]

Tell me more about this enigmatic Monty Montgomery

[@Miguel_Noether]

Some heroes don't use capes.....

[@psychomartian12]

What are 1/2 and 1/4 bits? How can a signal level be represented by less than a bit (1 and 0 )?

[@TangerineTux]

It’s not that it’s represented by less than a bit, it’s that its amplitude is less than that corresponding to the interval between two whole values. So you can have a digitised signal that alternates between 0 and 1 (or between [-1, 0, 1]), consisting of a “true” signal with a lower amplitude than that + some broadband noise.

[@baumstamp5989]

daaaaamn hifi-companies hate this trick. just imagine how much money has been made by lying.

[@JFSOUL]

Great video!

[@Xayuap]

so this problem was solved in the 80 like forever, except the compression issue. ¿can it get better at compression with more sampling and more depth?

[@artysanmobile]

Compression issue? What compression issue? Clearly, you haven’t listened to a word of the original poster’s excellent tutorial.

[@Xayuap]

you say no thong compression could be native.

[@jessegrisham]

Digital data compression dosn't really work that way, if that's what you're referring to.

[@Xayuap]

I mean for lossy formats

[@aangtonio5570]

@@Xayuap Maybe. But I guess you're missing the point why "lossy" formats were made for. They weren't developed for hi-fi or music production in mind, but audio data transfer and streaming, especially for the average human being and consumer-grade electronics. For that, very complex algorithms based on well-studied psychoacoustic models are applied to your 24-bit, 192kHz WAV file to convert it in a simple MP3, AAC or Opus file which can sound fairly good _in those use cases._ And they're becoming more and more efficient since the 90s (just compare between a standard 128 kbps MP3 file made with a 90s encoder, and a 128 kbps AAC or Opus file made with a state-of-the-art algorithm of today). For hi-fi, music production and personal storage, you can rely on a lot of lossless formats (some of them can compress up to the 50% of the source file). So, the "compression issue" today is a nonsense.

[@d4t4b4s3f4c3]

THANK YOU but what about the differences in the spectrum of harmonic distortion from analog vs digital amplification or recording?

[@ajcohen100]

I think I prefer the flat noise shape better. Over time the high hiss is more annoying.

[@teknolojigundemi]

Well I dont understand fully so basically 16 bit 44.1 khz is enough and we cant hear any noise related ADC-DAC conversion? Because it is so low? or something else.

[@stevenmsaxe]

Thank you for this explaination. It is greatly appreciated. It's not so much the bit rate, but the Digital to Analog Converter (DAC) at the far end tha makes the differentce. The stepping wave represenataion you show comes from a simple DAC circuit. You get a sample, it outputs the voltage that sample represents. A cheap DAC will just stay at that level til the next number arrives. Then to make the analog wave smoother it adds filters that get rid of the frequencies this stepping creates. The type of filters is somewhat dependant on the sampling rate. The time between samples is what causes the noise on these simple DACs. You essentially have a whole lot of square waves that have components of the sampling frequency, and some other stuff. Good filtering shapes the stepping wave to a a plenty good enough, or even a plentyier good enough analog wave that is very near the original signal. As with all engineering, many many variables were considered to get to the standard CD sampling rate. THank again.

[@thewhitedragon4184]

I'm not sure it's fair to say "A cheap DAC will just stay at that level til the next number arrives". The filter is a necessary part of the digital to analog conversion, and holding a number until the next number arrives, know as a zero order hold, is just a consequence of the fact that the output contains a capacitor that gets charged to that voltage level._(and even if we don't put a capacitor explicitly there is always parasitic capacitance of wires, traces, parts, hell even between the output point and the earth itself)_ We can't really get a series of dots into our analog filter as an ideal DAC would because there can be no device that can output a voltage which is a dot in time.

[@stevenmsaxe]

​@@thewhitedragon4184 Perhaps I should have said simple instead of cheap. Sorry. My point is that it is impossible to recreate an instantanious voltage in a DAC. You say you don't know where the step signal comes from. It comes from the DAC trying to recreate the original signal.

[@thewhitedragon4184]

@@stevenmsaxe Yeah the stairstep "effect" comes from the fact the we charge the output capacitor and it retains that voltage until another charging cycle comes. This essentially is a another sin(x)/x looking lowpass filter around the signal with some phase shift at the output

[@stevenmsaxe]

@@thewhitedragon4184

[@JasperPeters]

Yeah no that's just wrong... Digital signals (serial, ethernet, hdmi etc) usually switch between two states. They'd like to stairstep between the two voltages. But if you actually look at the voltage you will discover why no matter how simple or cheap a DAC is the output will not look like a stairstep. DAC's can be higher or lower quality, but they will never produce a stairstep.

[@rodriprat]

great master video!, wich software is used in the tablet?

[@user-zj4tw8in9w]

Спасибо

[@RandomEngineer69]

He is leaving out electrical engineering stuff for simplicity, this is an answer to those that still do not understand why it is not a step signal: A modern OPAmp does output a reasonably perfect step signal in khz use cases (assuming the OPAmp has a bandwidth of 200Mhz). But before a OPAmp output becomes a actual analog signal, it will also pass through a band filter(a combination of low pass and high pass filter circuits). This will limit the frequencies in the signal, making it lose its original shape, becoming “distorted” from a perfect step signal. Since we can only hear 20 to 20khz, when we record a sound, we also use a 20 to 20khz band filter before converting to digital (this also filters unwanted background static noises). The filtering on both sides will result in a “information compression and decompression” effect, so that as long as the interval for each step signal’s steps (equal to sample rate) doubles the highest frequency of the input filter’s cutoff frequency(in this case 20khz), the output filter’s distortion will perfectly “distort” the step signal back into the original filtered recorded signal. Yes, there is a-lot of googling needed to understand that, which is why he left it out. I think the misconception here is more of what defines a sound frequency(why not call a 60hz sine wave a 30hz “double sine” wave?). When we say humans hear 20 to 20khz, it specifically means pure sine waves. Anything else that is not a pure sine wave is just a bunch of different frequency sine waves stacked. For more information, google Fourier transform.

[@artysanmobile]

Jimmy, you are simply insisting on things that do not exist and using some very limited knowledge to support your completely erroneous theories. Nobody should even read your comment. It is a set of stair steps into an oblivion of ignorance on the topic, obscured by irrelevant technical language of topics you don’t understand. Just sentence after sentence of complete nonsense.

[@hannes7695]

You don’t need to know any electical engineering to answer the question about “why no stair steps”. It’s sufficient to know that there is only one correct mathematical solution when you go from digital samples back to analog, which reproduces the original signal perfectly. It’s not a quirk of how electrical components work or how some DAC implementations work like you seem to believe. It has nothing to do with filtering or smoothing. The filtering of half sample frequency before sampling is used to avoid distortion from alising and has nothing to do with human hearing or ”background noise”.

[@RandomEngineer69]

Sorry for the confusion, this reply is directed to those interested in making their own DAC from scratch. I noticed my reply isn't under someone's question as it was supposed to... That being said, thanks for taking your time to reply, I have learned alot from your kind feedbacks.

[@Pinpadprompts]

So many vinyl fetishists are fuming watching this

[@janbaca6075]

I've only understood 16 bits of this video.

[@extrabigmehdi]

The dither part was a bit too technical for me, but otherwise everything was clear.

[@DavidLindes]

Nice vid you did, way back when (I see it was 10 years ago). Dunno if you still have any interest, but I'd be curious to hear your take on Tom Scholz (of the band Boston)'s take on "phase angle distortion", as discussed about 50 minutes in to v=h7-w0s2IX3s -- a bunch of the stuff he said I was immediately like "yeah, but that's not how it works -- see Nyquist Theorem"... but I don't grok the phase angle distortion thing well enough to know -- is he just wrong about that, too? And if so, what explains the differences he believes he heard (may be unanswerable, but if you have thoughts...)?

[@DavidLindes]

P.S. My not-fully-thought-out theory is that phase information would fundamentally be higher-frequency information than the nyquist frequency, and yet if the phase information between different base frequencies somehow matters to how the human ear and brain perceive things, then indeed doing the band limiting needed for a 44.1KHz (or 48KHz or maybe even 192KHz) sample rate might still be perceivable with certain sounds -- that are combinations of certain base frequencies but at different phase angles. But, I haven't run the numbers and I don't trust that I'm right. It's just how I'm trying to make sense of what he's saying, so far.

[@gabrielbenitez6975]

So I don’t need to dither?

[@ky8920]

at 18:21 isn't the overshoot unavoidable even with infinite sine waves due to the discontinuities?

[@clehaxze]

It is avoidable with infinite sine waves. But only at infinite sine waves. As long as you add up a finite amount of sines, they will be there. (Source: My calculus textbook)

[@dlarge6502]

If I understand what you mean by overshoot then no. You need infinite bandwidth, nothing has infinite bandwidth. You need infinite bandwidth everywhere, including in the ear which is impossible.

[@chaoticsystem2211]

Would you even be able to notice the steps if they were present?

[@soloperformer5598]

That's why you need the MISSING distortion analyser.

[@dlarge6502]

No, your ears wouldn't be able to hear them.

[@musicien20011]

IMHO, the D/A or A/D converter likely works based on some Fournier's transforms with sine as base function, so the undegraded output based on single sine wave input is not a big surprise. It would be more interesting if we sample a real life sound signal which is not necessary a smooth function and see how the whole process goes.

[@ortzinator]

There's no difference

[@robertendert8944]

If you don't understand the fact that music is built up of multiple sinewaves of varying frequency, amplitude and phase, then it's end of story: you will never grasp it!

[@netoe]

I found that comparison to tape hiss a bit arbitrary. besides that great video.

[@kaffimann]

While I agree about the information presented in this video, and I agree for the most part that "16 bits ought to be enough for anyone" ;-) But I do think it's important to note that if you render a specific project using 24 bit depth there are some sounds that will simply not exist if you render the same project in 16 bit, so bit depth may have some significance at least for the rendering of a project. And the aspect of signal chain post processing and filters after the DAC are completely ignored, for my own part I am completely unable to hear the difference between 48kHz sample rate and 96kHz sample rate in a recording, but higher sample rates do have some merit in more advanced signal processing in the sense of it being less likely to tamper with the audible spectrum. Another aspect is that most real-life 16 bit DAC's have a noise floor at around 85-90 dB, not well above 100dB. There is a difference between theory and the end product. That said, I do not think there is much of an advantage getting much more than perhaps 18 bits of resolution or so even for the most eccentric of us, but if we are looking long and hard at a well made and reliable 24 bit DAC it will reveal that about 18 bits is more or less what you can expect from a quality 24 bit product all things considered. So to get true 16 bit performance you actually need a well made 24 bit dac.

[@kaffimann]

@@nicksterj Which is fine, though I don't really care for dither, but this is completely beside the point I am trying to make: The entire video blatantly ignores the fact that most 16 bit DAC's have a THD+N around 90dB or worse, which is why I am saying you need to have a very good quality 24 bit DAC to get PROPER 16 bit in your signal chain.

[@kaffimann]

@@nicksterj Agree 100% about the actual bitrate of quality soundcards, exactly as I wrote in my 1st post.I have absolutely no idea what kind of players you are talking about, my personal experience is limited to various external soundcards and balanced signal chains. Only thing I will say is that there is a significant difference between a 90dB THD+N and a 105dB THD+N signal chain. There is a slowly increasing number of DAC's able to perform better than 110 SINAD, biggest advantage is lower noise floor IMO, especially with high sensitivity speakers.

[@ЛевиАкс]

спасибо! супер!)

[@CuriousPassenger]

Those guys who say 'I want 192khz gor a higher audio resiluiton, 41100 is not enough for me' tho..

[@debrucey]

Whoever said nerds can’t be charismatic af

[@arthursgarage6550]

Something I'm a little confused on is how the signal is smooth, A DAC, from my understanding at least, can only output a finite number of analog values, so shouldn't there be a step function? Is the method of conversion significant?

[@mattymerr701]

​@subliminalvibes so it is because audio DACs are different to general purpose ones, or that they all approximate fixed voltage levels?

[@MostlyPennyCat]

I'm no expert, I'm not even a talented amateur, but my understanding is: A DAC is an _analogue_ device. You can model an electric signal as a weight on a spring. The DAC output voltage pulls on the spring, the weight movement is change in output voltage (the velocity of the weight is current, it allows down at a rate dependent on capacitance) You can "jump" from one voltage to another all you like, but the spring weight is analogue, it can only move through every voltage to get from A to B.

[@MostlyPennyCat]

Or, a DAC does not have discreet _output_ voltages, it has discrete _input_ voltages. And you can't listen to the input side, because that's just data, it's imaginary. You listen to the output, which is analogue, because _you_ are analogue.

[@arthursgarage6550]

@@sub-vibes I did watch the video, twice, I'm just confused on how a DAC creates a smooth signal is all, I may have misinterpreted the video and if so I'd be happy to hear where I went wrong. I'm not trying to spark an argument, I'm just confused about something is all.

[@binguloid]

@@sub-vibes related to sound, a speaker won't jump instantaneously between values; it still has inertia. It will have to accelerate for any change, meaning it will constantly be moving at any point between 2 samples, and will overshoot during big jumps.

[@vegettoblue8705]

Does he know ?

[@vihakurjategija]

So... your DAC does not matter?

[@jamessun7851]

The DAC does matter. His demo would be a bit different if he use non oversampling DAC

[@johnbarthol6493]

Hmm. At 5:51 I see the cable from the analog oscilloscope running under the signal generator but I see that the signal generator has a splitter. I'm thinking that one output from the signal generator is running to the scope and the other to the digitizer. Why else would the signal generator need two separate outputs? I'd actually like to see the cabling on this setup because it looks rather fishy to me...

[@KeithMilner]

There's nothing "fishy" about it. If you think he's, somehow, cheating, then show how he is: build your own setup to demonstrate where he is wrong and what the effect of doing the experiment correctly is. It's called peer review and repeatability, and it's what Science is all about. This is a relative cheap and easy setup to replicate. Debunking the above video should be easy if Monty is gaming the system in some way. The fact that, after 10 years, no-one has, speaks volumes. I'm sorry, but making claims about this looking "fishy" just comes across as someone butt-hurt by the video who can't accept the facts.

[@OMNI_INFINITY]

*Wait...each sample represents a voltage level on the speaker and is held for the sample duration. That IS a stair step pattern. So obviously the DAC (or some other interim processor) is doing the smoothing.*

[@thewhitedragon4184]

@@karmac0ma The lowpass filter at the DAC's output is doing that. I'm not sure there are dacs that don't include that. You literally need at least a single resistor and capacitor minimum.

[@karmac0ma]

@@thewhitedragon4184 youd be right

[@tz4601]

That is literally what a DAC does. Takes digital signals (samples) and converts to analog (not samples). The voltage is not "held constant" at a speaker. By the time the audio signal reaches the speaker cables, it's already been transformed into an analog (smooth) signal.

[@OMNI_INFINITY]

@@tz4601 Ah. How is the smoothing done? With capacitors and inductors?

[@DrBovdin]

Also, remember that any speaker is itself a filter, both electrically and mechanically. So even if we were to produce a “perfect” electrical square wave, once it is fed into whatever physical speaker you choose it would be filtered first by the coil, piezo element, or whatever you choose as an actuator and then as well by the transduction into sound waves. However, due to the infinite series of odd harmonics needed to construct a mathematically fully perfect square wave, the energy content of such a signal would also of course be infinite. That’s why an ideal square wave is unphysical.

[@EJBanks]

this was absolutely riveting. i tried to click off of it so many times...

[@cooksoni.a]

"The stair steps were never there" **Shows a zero order hold dac that has stair steps** (which was a very common dac in old cd players and is everywhere even today in random shit) unless you have a brick wall filter you're going to get some aliasing, and no circuit that uses a zero order hold dac is going to be a brick wall, let alone anywhere else really. Plus if you just reduce the sample rate or bit depth on anything it's going to show you stair steps, thats what the sound of aliasing looks like in the time domain... I'm not saying that any of this is relevant for listening to CDs, you're not going to hear it, but to say that the stair steps "were never there" is at best misleading. That's what aliasing is lol. Additionally, there's a reason why you should use oversampling on plugins that increase harmonic content like distortion. Going without it is begging for aliasing. If you've ever tried to make an FM synth, you'll certainly know you cant just run the DSP at 44.1 and expect it to sound good, especially when you add in feedback. so sure for listening formats, but this guy made it sound like it doesn't matter at all

[@nicksterj]

It's not misleading. The digital signal itself contains nothing like a stairstep pattern. Such a signal can _only_ exist on the analog side of the conversion - and he specifically mentions that zero-order hold is "an important part of how some digital-to-analog converters work, especially the simplest ones." So the stairsteps aren't part of the digital signal, they're created as part of the conversion process. And the most common type is sigma-delta which does not use ZOH, so no stairsteps at any point in the conversion. Hope that helps!

[@cooksoni.a]

@@nicksterj Yes so the stair steps are in fact there after the conversion if the signal aliases and it's not filtered out. Because again, aliased tones look like stair steps on an oscilloscope.

[@nicksterj]

@@cooksoni.a Right, but the imaging frequencies are created during conversion-they are not part of the digital signal. The myth he's trying to dispel here is that a digital signal "looks like a stairstep." In fact that's just a plot of sample values which have already been converted from the encoded numbers in the digital signal. A digital signal has only two voltage levels, representing 0 and 1.

[@artysanmobile]

How many times do we have to debunk this silly myth? Honestly, decades ago this was proven beyond any doubt. Put one of your very favorite commercially released CDs on and ask anyone to describe this imaginary distortion. If they fall for the bait, ask them to explain why they are right. Be prepared for lots of irrelevant nonsense.

[@artysanmobile]

@@josephgleespen1433 You’ve summed it up perfectly. Scanning the comments reveals a stunning number of people who continue to insist there’s still some hidden magic resolving their cherished stair step theory. They mention Fourier transforms, which have nothing to do with the topic and are guaranteed to be even more mysterious to them. I don’t even know where to start with these people. That perfect sine wave coming out of the D/A converter is somehow not convincing enough that the very bright people who tackled the issues of digital recording of music 60 years ago knew what they were doing. A tiny bit of knowledge is a scary thing.

[@bkik5]

None of this addresses the fact that smaller sampling and bit rates amount to lower resolution sound - or images, as the case may be. The complexity of sounds with all of the harmonics and other nuances sounds better at higher resolution, regardless of what this accountant says.

[@qmanol]

If noise is below the threshold of hearing, or below the level that is generated by the analog stages, then there is no value to increasing bitrate, as all it does is reduce quantization error, and thus noise. If frequencies are higher than can be heard (ultrasonics) or reproduced by the analog stages, then there is also no reason to increase the sampling frequency or change the bandlimiting. 192/96KHz and 32-bit sound is very useful in the recording, mastering and processing stages, but for audio delivery to the listener, 16-bit/48kHz are for all practical purposes at the limit of human hearing.

[@bkik5]

@@qmanol - With all due respect to his sine-wave machines and what they're worth, this is a gross oversimplification of what increased resolution actually means to sound. Bloody accountants thinking they have a clue about nuance.

[@bkik5]

@@nicksterj I'm no engineer but I call BS on that. It is absolutely related to resolution. Even if it's also related to things like THD and dynamic range. And no, it's bullisht to suggest that CD is 'good enough'. You guys just want to sound like you have some sort of inside scoop. You're basically reverse snobs. You're being ignorant.

[@bkik5]

@@nicksterj - Yes, my technical objection is the gross oversimplification of sound resolution to what his meters show in graphic form. He claims that if visible sine waves on a screen are smooth, then greater resolution has no effect except with noise floor. Rubbish. And the ground breaking statement that 'the staircase is actually a series of points'. Duh, really?

[@bkik5]

SonicScoop is another know-it-all talking out of his ass for clicks. I've been listening to hi-res vs standard res for months and I could pick out the hi-res every time. Similar to the audible improvement between shitty mp3 vs CD.

[@scottmarshall1414]

Well done, but I'd like to challenge a few of your claims. FWIW I've been working with digital and analog audio for over 40 years. Looking the same on a scope doesn't guarantee they will sound the same to our ears. Our ears are more perceptive of certain things than our eyes. I'm currently working on a project where I can hear problems in the sound that are invisible on my scope (100% analogue equipment and signals), but the only valid way to be sure if two sounds subtly differ is with properly conducted double and triple blind listening tests. Regarding your lollipop trace, this may be meaningful theoretically but doesn't show what's actually happening in the circuits that perform filtering, A/D, and D/A. A good A/D would not take an infinitely short sample of the audio stream 44,100 times per second. That would amplify noise. What's actually happening is the signal for the full sample time interview is averaged to produce the digital representation of the voltage for that sample. When converting back to analogue (if digital interpolation is not used) the analogue sample is output by the D/A for the full time period of the sample, then analogue filtering is applied to hide the stairstep. Now, at 16 bit samples 44.1 thousand per second, it's not surprising the artifacts are hard to hear and hard to see on a scope. However, you're not testing this at extreme audio cases. A low frequency at a low volume, for example, would more likely manifest the artifiacts of digitization -- not a medium frequency at high level. Finally, the noise that came from A-D-A tests was from the dithering, not the sampling. Without dithering, some really nasty aliasing can appear that is comparable to stair-stepping in digital images. Again, only significant with extreme musical program material, not ideal sine waves. I could write a whole book on this but that would be too much for a social media posting, where typically only the first sentence is read. All my best wishes!

[@gundabalf]

the only way to be sure is not scientifically accurate objective measurements, but a subjective test, uh huh, ok

[@scottmarshall1414]

@@gundabalf OK, Mister Sarcasm. There are imperfections in audio you can measure with instruments but can't hear, and those you can hear but may not measure with the instruments you happen to be using. Properly designed double-blind listening tests make the subjective become objective. What we perceive is, in the end, all that matters in sound reproduction

[@thewhitedragon4184]

I disagree with most of what you said. The lollipop trace is not just meaningful theoretical, but actually is just how one should view and treat digital data when processing it. It's a series of dots when it's digital. The point of an ADC is to give out our set of dots and the DAC is to take that set of dots and turn it back into the exact continuous line for which an ADC would give that set of dots, meaning a DAC doesn't just interpolate them. Here are some other points of contention I have with what you wrote from a technical point of view: 1) An ideal ADC would do the exact thing you said a good ADC wouldn't do. An ideal ADC would take and infinitely small time sample each period. That would just be a multiplication with a dirac comb. And an ideal ADC wouldn't need a quantizer. The problem is we can't make circuits which are just multiplication by dirac combs, we can make circuits that take a very small time to sample but not infinitely small, and we can't store a number with infinite decimal points, we'd need a bit depth of infinity, so we round. That rounding to our bit resolution of 8, 16, 24 or 32 adds noise to our signal that is equal to the difference of the unrounded signal and rounded signal. That's because that rounding error still has some energy so it manifests as noise in our quantized signal's frequency content. This noise is added on top of any noise the quantization circuit itself adds to the signal path. 2) Analog filters aren't used to hide the stairstep effect, they are an integral part of digital to analog conversion. Sampling a signal in the time domain is achieved, ideally, by multiplying the signal with dirac comb signal. That gives us a series of dots. In the frequency domain, this multiplication becomes convolution with a dirac comb _(the fourier transform of a dirac comb is a still a dirac comb)_ . This convolution makes copies of the original signals frequency content centered around the sampling frequency. So if you could have a circuit output a perfect set of dots that go to a lowpass filter, you'd get a perfect reconstruction of your signal. 3) I don't know what you mean by artefacts here. Real adcs and dacs introduce distortion from finite sampling times, quantization and the zero order hold sure but so does recording to an analog medium as the needle can shit or ambient noise can make on your tape. When you reproduce sound from a recorded medium it is always slightly distorted I don't get the digitization part. 4) Dithering is the only point I agree with you on but even then not fully. Dithering is deliberate noise added to the signal to preserve detail when reducing bit depth while sampling noise is inherent to sampling, I didn't choose to add it. Dithering is used because you record and mix with 24 bit audio and then export to 16 bits so to make sure your song doesn't become bit crushed you dither. This applies to sine waves too.

[@scottmarshall1414]

@@thewhitedragon4184 while the lollipop trace is theoretically interesting, it doesn't represent what the hardware is actually doing, which is all that matters in the end.

[@miroslavzderic3192]

@@thewhitedragon4184 damn spitting fatcs here

[@TheDudugomes77]

não estou compreendendo porra de nada que o cara fala, será que não fala Português

[@OMNI_INFINITY]

Any here can explain his supposed claim that tape has such a small bit depth? Really seems as though it's a video from a digital audio propagandist, so there is some skepticism. I have made analog synth emulation apps and it was a lot of work to get those to sound as high resolution as the analog edition.

[@thewhitedragon4184]

He's talking about tape hiss. When recording to a tape, you're physically blasting it with magnetic radiation. There is also other sources of magnetic radiation in the wild, namely the earth's magnetic core so you get noise on your recording added other then the noise already present at the signal. Your analog synth app honestly has very little to do with this other then the fact that if you plug a real synth to speakers and digital synth to speakers they can just make different voltage levels. Edit: Synth to speaker has no signal processing path, other then the frequency response of the speaker, potentially amp but let's assume that's flat, and like the impedance of the wire acting as and attenuator and like MHz filter. Your app has a DAC and it's noise and distortion to account for.

[@jonah1976]

He's talking about people who used their bookshelf stereo system to record music to tape from vinyl, FM, or CD, which were universally terrible. Automatic level control meant that the signal was always too low. No Dolby noise reduction. And people buy the cheapest tapes they can find, not chrome or metal. The next step up is using a dedicated tape deck recorder, which did sound better, but most people didn't use that.

[@OMNI_INFINITY]

@@thewhitedragon4184 Saw now. Thanks for the good info!

[@OMNI_INFINITY]

@@jonah1976 Ah.

[@dlarge6502]

He demonstrated that bit depth effects where the noise floor is. A CD with 16 bits has a noise floor down around -96dB or so, which with dither can push that down further to around -120dB. However tapes have a noise floor way way higher than that, typically if you were to look at the dynamic range of audio cassette in terms of digital bits then you'd have only 5 bits of depth. He was trying to highlight that the *quantisation* *noise*, noise added to a digital signal due to rounding the sample up or down to the nearest sample value, was so low as to be inaudible. This of course ignores any other noise added, say from a noisy microphone or preamp. Now on cassette you can go a long way to reducing the noise by using an erase head that records a high frequency tone to the tape before the audio is recorded. This makes the noise much less noticeable, plus with Dolby it can get even better. But still if expressed in bits, it's far short of what digital sampling can do. Assuming you eliminate other noise sources as I said.

[@soloperformer5598]

Why don't you add a distortion analyser to the set up and show the whole truth?

[@sabiro2315]

The spectrum analyzer does show harmonic distortion and noise, and was used quite extensively in his demonstration of bit depth and dither. I'm not really sure what else you want here.

[@soloperformer5598]

@@sabiro2315 I wanted to see the output on the instrument I trust, the HP spectrum analyser.

[@maxrainer804]

​@@soloperformer5598 it IS an HP spectrum analyzer

[@soloperformer5598]

@@maxrainer804 I know it is which is why I said it was the instrument I trust. There wasn't much output displayed on the HP spectrum analyser even thought the signal flow was from left to right.

[@ortzinator]

​@@soloperformer5598what will a distortion analyzer show that a spectrum analyzer won't?

[@jimmoss9584]

I can't believe this. He Eyeballs the output and says, "See, they are the same". what a scientist! Goofball is more like it. He could have bought the next level up HP audio analzyer. He would not be stuck with high Z. My product clients would not have been so easily fooled.

[@joshhunsaker]

This man is a genius. I can't tell you how many producers and audiophiles have no idea what they are talking about as it relates to the topics in this video. Thank god for those with actual engineering capability.

[@LRCProductionsTV]

I forgot to thank you guys for this video. I notice all the hard work that went into it, so thank you very much.

[@FL_STUDIO]

NOTE: You can't hear dithering under normal listening conditions. It does not impart anything 'obvious' to a recording other than replace one very low level distortion (quantizing error) with another very low level noise - 'hiss' of varying tonalities depending on the type. Dither is so quiet in order to hear it you need to crank the volume on the audio and the passage being monitored must be using only a few bits resolution (i.e very close to the noise floor).

[@HBStone]

Even though I knew the stuff in this video I loved every awesome minute of it. I would watch Monty talk about audio for an hour a day for the rest of my life.

[@FL_STUDIO]

They are interpolated values from the interpolation process. Interpolation wasn't really covered much in this video. However all those lines drawn through the sample points are exactly that...interpolation functions.

[@FL_STUDIO]

Similar. Tape Bias reduces non-linearities at low signal levels in tape (a form of distortion). While dithering replaces low level quantizing-error noise (a form of distortion) with (less objectionable) hiss (of varying flavors depending on the dither type).

[@FL_STUDIO]

Yes you can. Put the patterns in the Playlist. OR you can trigger them in Performance Mode

[@FL_STUDIO]

It's likely Live vs Rendered interpolation settings are the cause here not dithering. In the video, Monty makes the case that dithering from something higher down to 16 Bit is 'almost' inaudible. The dither effect is likely to be Just Noticeable, under ideal listening conditions, not something that would be immediately obvious.

[@messengercrow]

the video is recorded stereo, awesome, you can tell which side he walks to

[@catoffline]

Awesome show! And the text is very easy to understand for non-english-speaking people like me. :) Thanks so much!

[@DJignyte]

FL Studio is just a DAW, a platform or foundation for your VST's, effects and samples to play upon. What I think you're noticing is that when FL first starts up all values are set to the same where as other DAW's like Logic and Reason have native plugins that are already optimised and adjusted to sound nice right off the bat. I'm not 100% certain of this but I think that's all it is.

[@joseberroa4935]

In the past few months I had the problem with some kicks and basses from Harmor or after exporting to .wav or .mp3. Were some of them had like and after ugly sound at the end. The after kick sounded bit distortion and some basses if they have reverb or delay same thing to. It took me months and repeated video tutorial from you guyz to fix this problem. Using The EQ and Maximus, the interpolation and dithering to fix all this. Believe me!!

[@xiphmont]

Yes, it's Lousy Robot's sound to go for an old overdriven fuzz-box sound ala OK-Go (with the interesting exception of the keyboard). The music was not processed for in the video.

[@EDUBUDAN1]

Simply FANTÁSTIC explanation of digital audio probablly Stereophile readers will killl themselves when they finally discover that the snob ultra HD audio technophilia makes no sense at all !!!

[@noreaction1]

also, different DAWs have different dithers, and some set theirs a certain way by default. You should check your settings in each daw, and make sure they are consistent to make consistent results easier

[@bsoundbeatz]

I already read it...i trust my ears and i know what i hear.Try your VSTs in standalone and hear the difference in clarity

[@5ilver42]

this was very interesting to hear explained! thank you!

[@FL_STUDIO]

All thanks goes to Monty @ Xiph.org we just passed it along :)

[@TheStreetest]

This is a GREAT VIDEO!!!! Thnks alot! the world of music,Sound,Signals, etc... is INFINITE!!!

[@zmix]

Intersample peaks occur after the signal has passed through the reconstruction filter, usually in the analog domain. They don't exist in the "data" per se...

[@FL_STUDIO]

Indeed. Should have said we were discussing 16 Bit audio.

[@OtraCitiesMusic]

Fantastic video. That was incredibly informational.

[@joseberroa4935]

Yes friend if you record a CD with 32bit floating it does that automatically not me. The sound stays the same. But if you save as. Liked I said loses crystallization that's all. I have done hundreds of tries during the years. This is why I love this video cause teach people into the right direction of recording audio the right way. Peace..

[@-_Nuke_-]

The "nyquist" frequency is always the half of your samplerate. Now if you are using 44100 hz the nyquist frequency is going to be 44100/2=22050 hz. Now any frequency you produce and exceedes the nyquist frequency (for example the 22050 hz) in a digital work station (like fl studio) it will of course not going to be audiable from any human but it will produce some other frequencies (audiable to us humans) to "alias" back and create distortion. The name of that distortion is called "Aliasing"

[@SCKleiner369]

Excellent video. Thanks for posting.

[@Reggmail]

Thanks for the info, correct, in most blind test no one could tell the difference....Lol. ''Waves'' have been emulating for years and do a fantastic job.