Excellent.

I can say exactly the same! I have designed and built a DSP guitar amplifier. I have seen a lot of videos and articles that are incorrect and / or misleading. This one is superb!

@@antoniomonteiro1203 How did you find the process of building that, Antonio? Were there any resources/instructions/information you found particularly helpful? Something I've been thinking about trying my hand at. I'm pretty handy with various projects, but an Amp would be new territory for me and I'm not quite sure where to start.

@@jackallenproductions There is a lot of information and I believe it is not practical to put it here. How can we communicate by other means?

I've been dabbling into DSP myself, and I'd like some more info about this. I understand the basics of harmonic distortion and I understand what aliasing is. But does this unwanted effect, of reflected frequencies, happen due to processing in the time domain or in the frequency domain? I haven't tried to make a saturation plugin myself, but I'd imagine I would simply limit each sample in the time domain, with a transfer function that compresses the curve towards the extremes, so that a sinusoid would start to approach a square wave. Surely this would add third order harmonic distortion, but wouldn't cause any aliasing, would it? Sure, if I kept on adding odd harmonic sinusoids, and went above nq frequency, then there would be aliasing, but why would I do that? As you sure can tell, I haven't actually done any non-linear processing (coding) myself... :)

Yes the differences were so subtle I don't know if I could pass a blind AB test.

Dude I don't care if the mix sounds good or nah. I just want to be able to brag about having 14 compressor on my vocal track.

@@michaelvenne9386that’s just dumb

@@BR_READDY225 yeah.

You are not right, some things actually are related to production and music producers to use tools right, among them, your CPU performance limits you to do things right......

this!

Meanwhile a student who is studying DSP algorithms discovers the world of music production (me)

And that person is you.

Lol it's funny I'm watching this before turning the light out and going to sleep.

Haha. Gold

I fell asleep watching this vid.

kzfaq.info/get/bejne/r8eZhpaXmLaRYoE.html ok, its not 10 hours, but hey, Merry Christmas

😃 good idea

Lux Elliott or my wife’s.....

this should happen. I want Dan's voice as an option for Siri.

Dude me2 - and imagine they would give you so much well articulated and informed information

@@LoveMeBack that cracked me up thanks

Then you mustn't have tried. There are tons of resources on the basics of digital signal processing. There have even been great introductory videos for people that cannot read, like xiph's "Digital Show & Tell" published in 2013.

@@xnoreq no need to rain on his positivity dude....he's just a happy camper!

@@xnoreq oooh aren't you the clever girl

Same here man! He also presents in a manner which I want to listen to and not fall asleep to. Even though his voice would be GREAT for audio books haha.

I know I'm going to get my ass flamed & I would not trust this resource for anything historical nor political, but what's wrong with this definition ? en.wikipedia.org/wiki/Sampling_(signal_processing)#Sampling_rate

EVER!

..and Nyquist frequency.Thank you!

it should be at least a blind test, when you see changin text 44 oversampled/48 bla bla, it tricks your brain and you "hear" difference. but it the reality there's no difference maybe (now you need to do a blind test yourself, or remain a fool). then go watch pensado's place where he, with a friendly smile on his face, swears that he can hear a difference in audio cables. also every single itl where he lies to people then every single mixing tutorials channels where there is not a second of useful info and then think

@Gonzalo: Definitely among the best "processor" VST's I've ever used, but then I've not really tried many (any?) others since discovering FF! I guess thats a good AND bad thing lol!

I was going to make my own comment but this comment sure seems like what I was thinking so there you go!🤗😇🤪

@@gonzalob3348 Yep, bought most of them a while back. They are really all one needs (and yes, I have UAD, Waves etc)

@@C0D3O Genuinely curious. What kind of lies does he tell?

Igelkotte Good point, thanks.

I can see this being relevant for slow motion video. I'm curious, what are some other areas where you would significantly downpitch as to where this would have a relevant impact?

@@thorwaldjohanson2526 basically in any modern electronic music production/sound design environment, for instance drum'n'bass and jungle producers have been severely downpitching/pitching/timestretching all kinds of sounds since the 90's, also the whole "chopped and screwed" hip hop sub-genre is BASED around slowing songs down for example. Furthermore, changing the speed of a sound is basically one of the easiest ways of transforming it and making it unrecognizable, hence giving you creative options.

@@thorwaldjohanson2526 Auto-tune. The way it works is that it cuts audio into the tiniest chunks and sort of copy-pastes them and stretches them to alter pitch, length and vibrato. With 44.1 kHz, it sounds weird very quickly even with very mild auto-tuning. Even just having the recording in 48 kHz makes a big difference in my experience, now you can get away with correcting pitch and timing without it being noticeable. 96 kHz allows for extreme stuff if you're into that, but if you got somewhat capable musicians and are only fixing a few off notes or beats, 48 kHz works well for me (in Logic's own auo-tune system).

@@Hamachingo Thanks for the responses guys :). I'm curious, are you usually recording in 192khz 24bit and then downsample if need be, or do you start directly with 48khz? Or do you record different instruments at different settings? And are there any disadvantages besides filesize, to record at the maximum bitrate?

I would love to see Fabfilter make a DAW! :-)

OMG - can you imagine a FF DAW?! Who says Utopia can't exist?

I think Avid/PACE is a strong enough motivator to move on. There's plenty of better DAWs out there already.

@@AndrewWilson1991 This is *just* a question…seriously, nothing antagonistic AT ALL. Don’t read into this as anything other than curiosity. Do you do this for a living?

@@manifestgtranswer: you definitely don’t need protools too make a living in music/mixing..stop the bullshit

What do you think about recording the sounds of a keyboard through its headphone jack? Thanks.

Yeah I read an interview with one of the engineers that was on the CD team. Phillips Sony, there was a Toshiba guy there as well according the interviewee. It was fascinating reading.

Not all plugins have oversampling built in. They generate (fold back) aliasing in the audible frequency area when working on lower sample rates.

my only issue with that saying, is that it used to be Behringer's slogan, and I mean early Behringer when they were REALLY bad haha.

@@Bthelick Was it? I had no idea :D I only knew their slogan "We Hear You".

@@rickbiessman6084 yup I remember it clearly i worked music retail back in them days. In fact the irony of both slogans is palpable. by "we hear you" are they referring to how they like to slap people with lawsuits when they are found out how Behringer infringe the copyright / patents of other manufactures I wonder? because they are certainly listening out for that hahah

IISGIIG … “The Golden Rule” (read like Andy Samberg from the end of 3-Way … … …)

@Agent К_видео The difference is subtle, and falls into the category of "ear training" for certain things, plus a LOT depends on your monitors and room characteristics (or your headphones). I'm used to listening for "how much clipping is too much clipping" because I use clippers a lot and push limiters hard. The aliasing artifacts and IMD in the 44100 version has some similar "dirt" and "grit" around the edges of the transient sounds in the mix (like with clipping), and the mids and highs feel a little more "congested". By contrast, the 48000 version has a more "open" feeling in the mids and highs, and not as much of that dirty/gritty edge to the transients.

@@Baphometrix Agree with your assessment and my opinion was that neither were necessarily "better" I listened to the comparison with different sets of monitors, and for some it felt like the 44.1 version with OS was slightly more mid/low-mid forward and it made it feel like a more focused mix. No doubt that the 48 one felt a bit wider/open, but that's not always what I want/need in a mix.

Music is basically just fancy wiggly air

@@DeanLawrence_ftw Now THAT is too funny. And true!

I hope one day to be as wise as you 🙏🏽

About these high res 'record' releases (I'm assuming digitized recordings of analog master tape playback then of course, just stating this because in common parlance the term record is often read as synonymous with vinyl plate prints); To deny that they are *meant* for consumer consumption (as opposed to what consumption?) is simply contrarian. You know about these releases, so you must know that their resellers have a consumer demographic. I consume, as a consumer - which includes every producer because none of us only uses music materials for work aka in your terms "producer production": just to source, process, stock etc. - I consume hi-res consumer content recreationally, like the vast majority of their target demographics. And I'm not sure if you're simply generalizing a part of this market, or if you're making a case against the release of hi-res digitizations of any music source material, for any recreational consumers. Certainly when taking what you're saying literally, that is what you are doing; but of course communication is too complex to know for sure. In any case, consider the consumption of hi-res releases, by a consumer who bit-streams it to a high quality digital-to-analog converter, going into an analog (pre)amplifier which easily handles the output signal of the aforementioned converter, ending up, in any circumstance which assumes no use of destructive ignorent elements in the signal chain between, playing back through high quality speakers or headphones. What about this is 'not meant' to be?

ready to put it on linear phase at maximum and max oversampling? :D

I'm sorry to say this with a delay of one full year. But are you aware that if you resample anything from 48 kHz to 96 kHz, the only thing you get is your files taking up twice the space they originally took, without any improvement in sound quality whatsoever?

absolutely this

No you didn't.

That was a masterclass!

It was indees ! But I’m referring to a complete mixing course or masterclass.

Dan has own KZfaq channel and started being more active recently, keep an eye on it ;)

TzzSmk watched all his videos on his channel😅

@@BBDRecordsStudio Whats the name of his channel? Definitely interested!

And people pay a lot of money for MPC60s and EMU SP1200s......

This video demonstrated that you will hear a difference, assuming you apply digital processing of the signal to a common degree. What did you think the point was of him advocating the use of functions which use 8x the project sample rate, featured in the distortion plugins?? It just also demonstrated that you're better off using higher sample rates on a per-process/plugin basis, to avoid using resources on processes which cannot benefit from it (meaning not setting your entire project sample rate, based on the highest denominator which is your most aliasing-prone plugin(s) because it affects the CPU power needed for everything which doesn't audibly affect the signal as well. This has always been the case in out-of-the-box processing, because every analog signal processor, such as the type of distortions he's emulating here, was already 'upsampled' separately, by way of being an analog process, while the drum machine that the band used, was not unnecessarily digitally upsampled at the source, or at any digital point in the chain that didn't benefit from upsampling. But you will certainly hear the difference when you apply a digital process that aims to recreate the type of analog warmth that these distortion DSPs are doing. He's literally explained to you that you want very high sample rates for that type of processing, just that it's handy to have this function available and contained within each plugin, which in effect is just a way of getting closer to the way out-of-the-box mixing and mastering is done, assuming the inclusion of analog gear, which is virtually any music that isn't chiptunes.

@@JohnnyNatrium Thanks for your reply Johnny. I got it and that was exactly my point. Using plugins upsampling.. and not considering the AD/DA when using outboard or recording audio:-)