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
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!
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?
? 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.
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.
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 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...
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.
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 !!!
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.
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?
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).
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).
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.
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.
@@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.
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.
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.
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.
Ok.Now tell me why FL studio dont sound as clean as other daws like Logic?I opened up a Vst in standalone mode and it sounded way better. I im a Hardcore FL Studio user and i just want the sound quality fixed.
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.
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.
@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.
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 RUclips 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 RUclips 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)
@@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.
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 ?
To be precise, dithering to 16bit. Dithering is very important, necessary & very audible for 8bit audio.. which is why it's now as pointless as 8bit audio has disappeared from even the cheapest phone.
Couldn't have said it better my self. If this is not the best video on the planet then I don't know what is :-) Finally a video that is better than porn woohoo! lol...
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.
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.
Yeap I own a pair if computers. My main computer with Emu 0404 sound-card and the other one with the latest soundlaster card. Both computers hooked up with my speakers Behringer truth 8203. And I can tell the difference. If you export from Fl 32bit floating and then use for example soundforge ans " Save as" to 16bit 44.1khz Will sound different. A little bit nosier or loose crystallization. It doesn't sound that crisp. Thats all im saying. Before I sell to Itunes I make sure my audio sound best.
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.
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
For example and I don't recomend nobody doing this. If you export 32bit floating, record that to a Cd. But never go to Soundforge or any other sampler to lower to 16 bits after exporting from FL Studio. Do that Directly from FL Studio. You will have the best Sound Possible always from Direct Exporting to your needs. Fl Studio has everything...Best Daw in the world.
Yo i'm going from logic (leaving apple) to FL demoing it right now on a PC i have,I need to know if you can let's say do a drum sequence on pattern 1 then jump to pattern 2 automatically or do i have to do it manually?
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).
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!!
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..
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
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
No friend jhejeje I dont truncate and I dont recommend that at all. is completely in backwards. I normally export my audio in 16bits 44.1khz I dont record in 24bits or 32bit.
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.
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...
"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 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.
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.
I show this to analogue purists and audiophiles on a regular basis.
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!
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
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.
This is so well presented it’s crazy!
daaaaamn hifi-companies hate this trick.
just imagine how much money has been made by lying.
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!
In 100 years this video will be a treasure
It already is.
It always will be...
Still is! 😂
I've never seen such a technical video explained in such a great way. Thank you, saved me a couple of readings.
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?
This is, by far, the best demonstration and explanation of this subject that I have ever seen. A true mic-drop moment!
? 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.
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.
What a GREAT explanation that is understandable about a misunderstood concept. Thank you.
All thanks goes to Monty @ Xiph.org we just passed it along :)
Impressive! Excellent presentation. Thank you!
Another one conned.
what is your problem?@@soloperformer5598
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
No, that's why we process in 32 Bit floating point.
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...
So many vinyl fetishists are fuming watching this
Your beard is awesome
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.
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 !!!
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.
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?
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).
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).
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.
Still the GOAT vid on the subject, hands down.
what a perfect video!
Kinda!
Tell me more about this enigmatic Monty Montgomery
Yes you can. Put the patterns in the Playlist. OR you can trigger them in Performance Mode
Whoever said nerds can’t be charismatic af
I've only understood 16 bits of this video.
When you do, post it in Looptalk so we can discuss it at length.
What you can hear is not the truth - watch?v=G-lN8vWm3m0
Those guys who say 'I want 192khz gor a higher audio resiluiton, 41100 is not enough for me' tho..
Indeed. Should have said we were discussing 16 Bit audio.
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.
@@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.
Some heroes don't use capes.....
Wow - great demos and explanations
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.
Such an interesting video and only 17,000 likes. Thank you very much! 👍
If they are relevant to our customers, yes.
the video is recorded stereo, awesome, you can tell which side he walks to
You made a mistake.
Google - 'Audio Myths and DAW Wars'
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.
TheWhiteDragon, whoever you are, you are one of my new favorite people.
great master video!, wich software is used in the tablet?
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.
See the video information
What analyzer software are you running on the laptop? Excellent debunk video!
Ok.Now tell me why FL studio dont sound as clean as other daws like Logic?I opened up a Vst in standalone mode and it sounded way better. I im a Hardcore FL Studio user and i just want the sound quality fixed.
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.
This video is so good and useful
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.
I think I prefer the flat noise shape better. Over time the high hiss is more annoying.
@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.
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 RUclips 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 RUclips 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.
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?
Compression issue? What compression issue? Clearly, you haven’t listened to a word of the original poster’s excellent tutorial.
you say no thong
compression could be native.
Digital data compression dosn't really work that way, if that's what you're referring to.
I mean for lossy formats
@@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.
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 ?
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
@imageline how is that?
To be precise, dithering to 16bit. Dithering is very important, necessary & very audible for 8bit audio.. which is why it's now as pointless as 8bit audio has disappeared from even the cheapest phone.
This video deserves to have MILLIONS of views. There's still so much misinformation going around about digital audio.
This is the best video on the internet
Couldn't have said it better my self. If this is not the best video on the planet then I don't know what is :-) Finally a video that is better than porn woohoo! lol...
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.
LOL check my page and you tell me if i do or do not know how to use FL Studio nut face.
THANK YOU
but what about the differences in the spectrum of harmonic distortion from analog vs digital amplification or recording?
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.
Yeap I own a pair if computers. My main computer with Emu 0404 sound-card and the other one with the latest soundlaster card. Both computers hooked up with my speakers Behringer truth 8203. And I can tell the difference. If you export from Fl 32bit floating and then use for example soundforge ans " Save as" to 16bit 44.1khz Will sound different. A little bit nosier or loose crystallization. It doesn't sound that crisp. Thats all im saying. Before I sell to Itunes I make sure my audio sound best.
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.
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
this was absolutely riveting. i tried to click off of it so many times...
The dither part was a bit too technical for me, but otherwise everything was clear.
For example and I don't recomend nobody doing this. If you export 32bit floating, record that to a Cd. But never go to Soundforge or any other sampler to lower to 16 bits after exporting from FL Studio. Do that Directly from FL Studio. You will have the best Sound Possible always from Direct Exporting to your needs. Fl Studio has everything...Best Daw in the world.
I can pass my class. Thanks god for this video.
Yo i'm going from logic (leaving apple) to FL demoing it right now on a PC i have,I need to know if you can let's say do a drum sequence on pattern 1 then jump to pattern 2 automatically or do i have to do it manually?
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).
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!!
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..
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
So would it be correct to say that dithering somewhat behaves like bias signal in audio tape?
I found that comparison to tape hiss a bit arbitrary. besides that great video.
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
No friend jhejeje I dont truncate and I dont recommend that at all. is completely in backwards. I normally export my audio in 16bits 44.1khz I dont record in 24bits or 32bit.
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.
I forgot to thank you guys for this video. I notice all the hard work that went into it, so thank you very much.
This is a GREAT VIDEO!!!! Thnks alot! the world of music,Sound,Signals, etc... is INFINITE!!!
very well made video. very well explained and narrated. good job!
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.
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...
The fight of truth vs. "truthiness" continues I see. Ah well. Only so many pseudo-engineers can be converted I guess.
hey imageline next version of fl studio change the gui to look like a roland d 50
Google, "Audio Myths and DAW Wars". Click on the ImageLine article that should be first on the list. Read it thoroughly.
"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 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.
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.