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.

[@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

[@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.

[@6sixtysix]

In 100 years this video will be a treasure

[@VictorVonBelmont]

It already is.

[@ghreghdehomeshrhamesh8639]

It always will be...

[@lewiswaddo5045]

Still is! 😂

[@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!

[@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)

[@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!

[@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.

[@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.

[@WindsurfingNelson]

Impressive! Excellent presentation. Thank you!

[@soloperformer5598]

Another one conned.

[@derpz_]

what is your problem?@@soloperformer5598

[@FL_STUDIO]

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

[@FL_STUDIO]

? Monty is referring to music distribution formats not music production formats. 32 Bit floating point is necessary for music production. 24/192 is not only unnecessary for music distribution, 16/44.1 being the gold standard, it may even sound worse.

[@cstefanile]

Still the GOAT vid on the subject, hands down.

[@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.

[@MarcRios-w2t]

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

[@FL_STUDIO]

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

[@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).

[@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?

[@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.

[@CisterJr]

what a perfect video!

[@soloperformer5598]

Kinda!

[@stephenyoud6125]

Wow - great demos and explanations

[@goldenretrogames]

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

[@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 !!!

[@FunkyELF]

This is the best video on the internet

[@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!

[@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

[@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.

[@ParboiL]

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

[@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!!

[@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...

[@messengercrow]

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

[@TheStreetest]

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

[@SomeoneOnlyWeKnow.]

This video is so good and useful

[@FL_STUDIO]

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

[@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..

[@FL_STUDIO]

No, that's why we process in 32 Bit floating point.

[@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

[@5ilver42]

this was very interesting to hear explained! thank you!

[@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.

[@OtraCitiesMusic]

Fantastic video. That was incredibly informational.

[@AvithOrtega]

An oldie but goldie

[@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...

[@SCKleiner369]

Excellent video. Thanks for posting.

[@FL_STUDIO]

Of course there is likely to be a difference and it's likely to be explained by monitoring levels. You are aware there is a Limiter on the Master Mixer track 8 associated with the default project? You matched the output volume of the Stand-alone with FL Studio? You set the same audio driver for both?

[@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.

[@FL_STUDIO]

When you do, post it in Looptalk so we can discuss it at length.

[@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.

[@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.

[@-_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"

[@kenclarke5966]

Tell me more about this enigmatic Monty Montgomery

[@EvanZalys]

It depends on what you're after. The classical moog latter filter is quite hard to replicate digitally, as it turns out. Analog instruments drift and distort and all the little nonlinearities make the sound more interesting, and any emulation has to account for all of these. Can you tell the difference? I don't know. OTOH if you're using something like an Access Virus Ti or something, that's digital internally, and a VST would do JUST as good a job. So the answer to your question is it depends

[@iamkieran]

Thank you for such an awesome video!

[@vazzed]

Thank you Imageline!

[@VincentRubinetti]

very well made video. very well explained and narrated. good job!

[@SpencerTwiddy]

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

[@solomonomar]

Your beard is awesome

[@joshhunsaker]

That's another area where it's easy to get foiled by technicalities however. Unless you're using plugins that respond dynamically based on input signal level, gain staging is effectively meaningless in 32bit float (until you export to a non-float format).

[@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. 🫵🏻🤡🤡🤡

[@baumstamp5989]

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

[@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

[@AndreasBecker-t2h]

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]

@@AndreasBecker-t2h 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.

[@27b-6Buttle]

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 ?

[@bsoundbeatz]

@imageline I'm not bashing FL studio,it's the best program out there.All im saying is that there is a problem with the sound engine.Please run the test for yourself and you will hear the difference.

[@whoiscm]

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

[@scottfaircloff9530]

thanks for this!

[@rafaelmarfil]

Superb! Thank you so much!

[@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.

[@joseberroa4935]

Forget Batman!! your my new Hero!!!!... This is why one of my strongest reasons why I use dithering when I'm exporting audio. I do it cause when I use 16bit drums samples. They don't match 32bit vst software sound. Dithering and interpolation is really important when you record vocals and use samples in a song. Sounds better!!! You can practice with drumaxx and drum samples and that will give you and idea of what I'm talking about. ps: remember to use Maximus best compressor ever.

[@Aceley]

this is the best show since "fun with flags"

[@vileguile4]

thank you for this video!

[@mark0nius]

Yes, but I was wondering if you'll be putting them here, since it's simpler to follow them.

[@Hurdstah]

this is brilliant!

[@FL_STUDIO]

If they are relevant to our customers, yes.

[@Miguel_Noether]

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

[@wild939]

i will always believe that analog generated sounds will always be more clear and better cuz its the truth and not about physics or math. its about what i can hear

[@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.

[@BLaQkRoWHaloSG]

More please!

[@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.

[@ZAza-nt5gd]

Good job

[@Yotrymp]

How about a video explaining the phase issues that come along with using equalizers. I've read about it, but I don't understand much about it.

[@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.

[@FL_STUDIO]

Google - 'Audio Myths and DAW Wars'

[@joshhunsaker]

Google, "Audio Myths and DAW Wars". Click on the ImageLine article that should be first on the list. Read it thoroughly.

[@ajcohen100]

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

[@FFamily713]

Disregard i figured it out,will be buying soon.

[@-_Nuke_-]

That was a great video !!!

[@SilexzStudios]

Awesome

[@jmdigriz]

What analyzer software are you running on the laptop? Excellent debunk video!

[@Komarovskimusic]

Cool cup! :) Thanks, great video! ;)

[@d4t4b4s3f4c3]

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

[@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.

[@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.

[@JFSOUL]

Great video!