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!
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
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)
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.
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.
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?
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.
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).
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.
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
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.
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).
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..
? 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.
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 !!!
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!
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!!
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?
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.
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.
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.
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.
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
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.
You know that what you propose would be illegal. Contact authorities and sue them or at least prove it or... keep accusations to yourself if your only goal is to spread uncertainties.
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
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.
@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.
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.
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.
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.
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.
@@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.
@@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
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.
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...
Assuming they want to know, which I doubt. This information in one form or other is rather easy to come by. But if you have spent 10000USD on "quantum ground conditioner", you don't want to know that you simply wasted your money. Sadly, fighting disinformation or outright conspiracy theories with knowledge, however accessible, is not going to change people's mind. As far as I found out about it from psychology standpoint, we rather look at deprogramming methods for sect members (where such knowledge would be a step, but late one), not a link to video.
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
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...
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).
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.
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
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.
@nicksterjNot sure if TH-cam 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. 🤨
@@AndreasBecker-t2h Not sure if TH-cam 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)
The datasheet of the Philips TDA1543 NOS DAC suggests supplying it with 4x oversampled pcm data (i.e. 176.4 kHz) and in the block diagram they put an external first-order low-pass filter with 40 kHz corner frequency. If you follow both suggestions, i.e. use the DAC as it was intended, this will result in pretty smooth output. Maybe if you look closely there might be some tiny wrinkles (far outside the audio band) but it will absolutely not resemble a stairstep. The NOS Audio DACs of old time were trying to achieve the exact same result as modern Audio DACs, a properly reconstructed waveform. They may not be quite as good at it as modern ones, but the only way to actually get a stairstep output is if you deliberately ignore how the DAC was intended to be used.
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"
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.
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.
Because despite having read 'Audio Myths and DAW Wars' you seem to have fallen victim to one of the 9 traps posted. If your stand-alone VST sounds 'better' then look for the cause there.
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.
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.
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”.
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.
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 ?
![ประสบการณ์ จ้าง Developer ที่ผมจะไม่มีวันลืม [ภาค1]](http://i.ytimg.com/vi/bzivN4KDg-k/mqdefault.jpg)
This is, by far, the best demonstration and explanation of this subject that I have ever seen. A true mic-drop moment!
The showmanship in this video is astounding
But misleading.
@@soloperformer5598how is it misleading?
And distracting
@@SamsungTshirt Its not, he just had to come in here and make sure you knew.
Not a chance, it's styled like a tutorial for elementary schools. I don't see how that's a problem.
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!
I've never seen such a technical video explained in such a great way. Thank you, saved me a couple of readings.
This is so well presented it’s crazy!
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!
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
What a GREAT explanation that is understandable about a misunderstood concept. Thank you.
In 100 years this video will be a treasure
It already is.
It always will be...
Still is! 😂
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)
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.
Impressive! Excellent presentation. Thank you!
Another one conned.
what is your problem?@@soloperformer5598
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.
This was one great tutorial!! Clear, concise, well explained! Thank you for your time and effort...it is greatly appreciated!!!
I forgot to thank you guys for this video. I notice all the hard work that went into it, so thank you very much.
I show this to analogue purists and audiophiles on a regular basis.
This is an awesome video presentation. Spectacular.
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?
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.
Yes you can. Put the patterns in the Playlist. OR you can trigger them in Performance Mode
All thanks goes to Monty @ Xiph.org we just passed it along :)
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).
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.
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
Wow - great demos and explanations
Still the GOAT vid on the subject, hands down.
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.
This video deserves to have MILLIONS of views. There's still so much misinformation going around about digital audio.
what a perfect video!
Kinda!
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).
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..
Awesome show! And the text is very easy to understand for non-english-speaking people like me. :) Thanks so much!
Indeed. Should have said we were discussing 16 Bit audio.
This is the best video on the internet
? 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.
this was very interesting to hear explained! thank you!
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 !!!
Such an interesting video and only 17,000 likes. Thank you very much! 👍
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
When you do, post it in Looptalk so we can discuss it at length.
Fantastic video. That was incredibly informational.
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!
This is a GREAT VIDEO!!!! Thnks alot! the world of music,Sound,Signals, etc... is INFINITE!!!
Excellent video. Thanks for posting.
This video is so good and useful
the video is recorded stereo, awesome, you can tell which side he walks to
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!!
An oldie but goldie
Thank you for such an awesome video!
No, that's why we process in 32 Bit floating point.
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?
very well made video. very well explained and narrated. good job!
Thank you Imageline!
TheWhiteDragon, whoever you are, you are one of my new favorite people.
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.
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.
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.
@@srpenguinbr Excellent and concise explanation. Bravo.
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.
@@FM-kl7oc Exactly, and since we don’t have infinite data bandwidth, the integrity would surely suffer.
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
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.
You know that what you propose would be illegal. Contact authorities and sue them or at least prove it or... keep accusations to yourself if your only goal is to spread uncertainties.
I can pass my class. Thanks god for this video.
Tell me more about this enigmatic Monty Montgomery
This explanation is perfect
But limited.
@@soloperformer5598 "BuT LiMiTeD"
Bet you are one of those people that use the "Your audio isn't distorted enough" argument. 🫵🏻🤡🤡🤡
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
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.
this is the best show since "fun with flags"
thanks for this!
@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.
If they are relevant to our customers, yes.
Your beard is awesome
Superb! Thank you so much!
daaaaamn hifi-companies hate this trick.
just imagine how much money has been made by lying.
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.
thank you for this video!
this is brilliant!
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.
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.
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.
@@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.
@@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
@@thewhitedragon4184
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.
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...
Assuming they want to know, which I doubt. This information in one form or other is rather easy to come by. But if you have spent 10000USD on "quantum ground conditioner", you don't want to know that you simply wasted your money. Sadly, fighting disinformation or outright conspiracy theories with knowledge, however accessible, is not going to change people's mind. As far as I found out about it from psychology standpoint, we rather look at deprogramming methods for sect members (where such knowledge would be a step, but late one), not a link to video.
That was a great video !!!
Good job
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
More please!
Cool cup! :)
Thanks, great video! ;)
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...
Disregard i figured it out,will be buying soon.
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).
Some heroes don't use capes.....
Great video!
excellent video, thank you very much!
But, it might have been nice to include the answer to the question "What is quantization?" :)
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.
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
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
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.
@nicksterjNot sure if TH-cam 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. 🤨
@@AndreasBecker-t2h Not sure if TH-cam 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)
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.
The datasheet of the Philips TDA1543 NOS DAC suggests supplying it with 4x oversampled pcm data (i.e. 176.4 kHz) and in the block diagram they put an external first-order low-pass filter with 40 kHz corner frequency. If you follow both suggestions, i.e. use the DAC as it was intended, this will result in pretty smooth output. Maybe if you look closely there might be some tiny wrinkles (far outside the audio band) but it will absolutely not resemble a stairstep.
The NOS Audio DACs of old time were trying to achieve the exact same result as modern Audio DACs, a properly reconstructed waveform. They may not be quite as good at it as modern ones, but the only way to actually get a stairstep output is if you deliberately ignore how the DAC was intended to be used.
Google - 'Audio Myths and DAW Wars'
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"
LP filters would like to have a word.
@@filipvidinovski7960 oh no! XD
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.
great master video!, wich software is used in the tablet?
What analyzer software are you running on the laptop? Excellent debunk video!
What are 1/2 and 1/4 bits? How can a signal level be represented by less than a bit (1 and 0 )?
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.
Awesome
See the video information
Because despite having read 'Audio Myths and DAW Wars' you seem to have fallen victim to one of the 9 traps posted. If your stand-alone VST sounds 'better' then look for the cause there.
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.
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.
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”.
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.
Those guys who say 'I want 192khz gor a higher audio resiluiton, 41100 is not enough for me' tho..
Google, "Audio Myths and DAW Wars". Click on the ImageLine article that should be first on the list. Read it thoroughly.
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 ?
He used analog test hardware to increase the probability it might convince people who think everything analog is better
@@MatthijsvanDuin Too bad he never tested it on a flat Earth :)