Debunking the Digital Audio Myth: The Truth About the 'Stair-Step' Effect
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- Опубліковано 20 тра 2024
- Learn why 16-bit/44.1kHz audio is just as good as high-res audio formats for playback (if not better)!
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This video was originally created by Christopher "Monty" Montgomery and xiph.org. The video has been adapted to make the concepts more accessible to viewers by providing context and commentary throughout the video.
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Part 2: ua-cam.com/video/VSm_7q3Ol04/v-deo.html
Watch Monty's full video here: ua-cam.com/video/UqiBJbREUgU/v-deo.html
Thanks to Monty Montgomery and xiph.org for making this information available with a Creative Commons License!
Some interesting insights into DAC reconstruction filters can be seen in the two application notes by Analogue Devices AN-823 and AN-837 for Direct digital synthesis applications but theory is similar.
A video describes the same details ua-cam.com/video/dD9HC1GThZY/v-deo.html
The original video featured here left out the reason that the stair step waveform is not visible on the output. It is because of the reconstruction/output filter after the DAC.
Denying the existence of the stair step is a little disingenuous though.
AudioUniversity, huh?? I’ve lost count at this point of the misinfo provided as opinion, and downright incorrect info offered as education. Too bad UA-cam just doesn’t care. But I do. I now know this channel to be a safe space for cr*p, home to clickbait nonsense.
@@nicksterj Were you directing the question at me? The glossing over of the fact that the output from the DAC is a stair-step before the reconstruction/output filter which is sometimes digital but traditionally analogue. The lolli-pops only exist on paper, in the digital realm it is just values. Once it is electrical it is a continuous curve (voltage or current) that will be slew rate limited but it will not be discontinuous. The stair-step is the electrical output you expect from an unfiltered DAC, trying to promote that it does not exist is not correct.
The higher db really helps w/hearing corrections & poor audio can't handle those levels :)
ECCO LA VOSTRA SINUSOIDE
I'm a professional sound engineer and technician in live theatre, with a background in electrical engineering. While most of this information was stuff I learned in theory, the real-world implications and applications of it are something I never gave much though to and always took for granted. That first experiment you presented was a huge "duh" moment for me.
This is an amazingly informative video. Thank you for making it and presenting the information in such a clear and concise way.
Yeah, knowing about Fourier transform and low pass filters is one thing, seeing it "live" is another.
The nifty thing about using Inductor/Capacitor filters is that they can do a breathtaking job of reconstructing anything vaguely within their resonance value.
This is something that pretty much all of radio communication has been built on since day 1 of it. Converting back from digital is really not that much difference then taking the intelligence signal off of a radio frequency, as far as the mixer/filter side of it is concerned.
Yes, got pretty much same feelings. I had always taken these for granted with my technical background (liking this explanation and demonstration though), but maybe for laymen with their misbeliefs, the simple remedy, without need for this long technical explanation, would be to just simply mention that the sample points are not stepped, they are interpolated.
Same, it basically just took everything I knew, and put it in a digestible format. That being said, I'll still stick with my 24/192 FLAC collection, but that's basically entirely for future-proofing. If I want to actually listen to something and have it sound incredible, I'll put on a record; and only there I know it only sounds "good" because it's actually horrendously imperfect. I think it evokes something subconsciously with relation to seeing a performance vs. hearing a pristine copy of it.
@@tomshotdogs6645 Yes, purely psychological component of the experience is entirely different can of worms.
There's a reason to use 48khz in movies/videos which has to do with the fact that it's a multiple of 24 which in turn means you get an integer number of samples per frame. Makes syncing video and audio much easier.
My experience is, it sounds better, especially the high end. Comparing CD and DVD side-by-side, the DVD is a little more "crisp".
The only correct answer.
@@vincentrobinette1507lol
@@vincentrobinette1507 'crisp' is a subjective thing in human ears. and human senses are not that good. the human mind doesn't work like we think it does. EQ on every stereo are there because human hearing is really not that great lol. the human senses, barely work. if you've seen an optical illusion, you know that.
@@GungaLaGungaI dunno, a handful of fleshy tubes that can detect oscillations of molecules on the order of nanometres with wavelengths spanning several orders of magnitude sounds pretty good to me.
As an audiophile I’ve always tried not to believe in things I don’t understand. I just experiment, listen to things and whatever my ears agree with is what goes. Many times I have preffered the sound of something cheaper, simpler or “less audiophile”. Or found no difference. The key is to pay minimal attention to what people are saying and just try things. Often they are just repeating things they hear or attempting to fit in with a group.
I am using lossy encoding whilst all the people scream FLAC. The amount of parroted nonsese is piling up to literal fairytales of untrue garbage. I play them my MP3 (!, the lowest denominator amongst AAC, OPUS, VORBIS and the likes) created with Lame 3.100 at V0 and they go like "Sounds actually pretty awesome". I tell them it's an MP3 averaging at 275 kBit/s and immediately they claim that they can hear "bad this" and "bad that", and that sound was "hissy" and how they rather want 800 kBit/s. I make an A/B switch on Audacity "showing" that there is no audible difference - nope nope nope, FLAC is the only way and MP3 is bad. Huh.... I even give them the WAV and all the "hissy" things are present in the WAV as well and not butchered by the encoder. Nah, dude, all bad. These people like to keep ultrasonic content and irrelevant audio information at all cost. . . . . .
Placebo effects are a thing, but not only with your ears, but also with your brain. Maybe analog to a kid that fell off of a bike: The knee hurts. Ouch, but fine. Same accident, same pain, but now the he kid sees blood on the knee. Waaaaaaaa!!!! Mama!!
In a shootout of high-end speaker cables a pair of wire cloth hangers won.
@@ZedekThat's the difference between audiophiles and audio engineers. Audiophiles rely on subjective impressions to judge their equipment, with all the human fallibility and bias that brings. Engineers have test equipment to measure it.
@@Zedek Yeah, back in around 2001 (I remember that year vividly because I did this at a unit I was living in when 9/11 happened), I had heard that LAME 256k CBR was supposedly so good that it was indistinguishable from the lossless source. I was running Debian Potato at the time and thought I'd try it out.
I started ripping with cdparanoia my favourite CD's and then converting those rips to LAME 256k CBR, to do some tests. I was doubtful, so as is human nature I was searching for unwanted, audible artefacts. I was shocked that these MP3's sounded so amazingly good and then suddenly I heard it! A warble in one of my favourite songs, which was not musical, seemed out of place and I did not recognise. I went back to the source CD and listened......... and there it was, that warble, on my pristine CD, of one of my favourite songs, that I'd never noticed before.
I checked other sources of the same song and what do you know, that warble was there too. I now cannot un-hear that warble when I listen to that song.
The ultra critical headspace I was in, while listening, caused me to find what sounded like non-musical artefacts in my favourite music, yet at the same time I was completely unable to find anything detrimental added by the LAME 256k CBR encoding process.
So yeah, LAME is a pretty awesome encoder and I understand it has likely improved even more over the past 20+ years and that there are some really awesome and ultra efficient open source codecs now (OPUS rings a bell).
@@ZedekDisparity of samplerate and audible lossy compression are not analogous at all, IMO. I can 100% hear lossy compression, but I have $8k monitors and a treated room. Just don't throw the baby out with the bathwater, there are 100% people out there who are not lying to you when they say they can hear mpeg compression. This is a vastly different conversation from the samplerate debate.
I remember seeing the original Xiph video almost 10 years ago now, and it fundamentally elucidated digital audio to me. Great to see it being shared beyond its original, as it didn't get nearly enough attention.
The best digital recordings are going to be the ones with the most talented mastering. That will have far more impact than high resolution audio. That said when doing mix downs the high resolution seems to be much easier to achieve that great sound.
me, too. Revealing
Note: 20KHz is the highest frequency a teenager can hear. Enjoy it while you can kids because its not going to last. As you age the limit drops. For most adults the CD rate is not just good enough, its way beyond good enough.
Most teenagers have already blown their 20KHz perception with their listening habits. At least we did in my day. This is more the range of what a 5 year-old can hear.
Yeap, can't hear the whine of a CRT monitor anymore. But, boy it bugs the hell out of those youngins to play some Nintendo on one...
Sometimes I wonder what my cats actually hear when I play a CD. (That's a lon time ago to be honest.)
@@ouwebrood497 It's going to sound like a telephone to them. The noise is all filtered out but so are the high frequencies that they can hear but we can't. If a cat had designed the CD instead of a human the sampling rate would have had to be around 180KHz because they can hear up to 85KHz.
nah, im still annoyed at the "rat" repellent, when i go to a certain store that uses it, its painful.
and im way past the teenage years.
Something not mentioned is that the raw DAC is followed by a low-pass filter at the Nyquist frequency. So, even if the stairsteps are there, it's at the sampling frequency and would get filtered out.
Was thinking exactly the same thing. This is correct !
Exactly. Thank you.
And you filtered out the high frequency sound, too. But anyway with powerful headphone all days long can't hear them anymore. a 44kh give you the capacity to produce 22kh square wave sound. To produce decent 22kh sine wave, you need a sample a least 8 time higher. You can use Bezier algo to reshape the sound wage, but it not sound an accurate sound reproduction anymore.
@@omegaman7377 you have that backwards. With sine waves as your basis function it's impossible to perfectly model a square wave without infinite sine waves. The Nyquist frequency is plenty good for sine waves because it IS the basis function of the model. You can apply more advanced techniques to reconstruct beyond Nyquist, but not in a general way, so it's not useful for audio applications. We can do this in image processing (though with some additional information which is kinda cheating)
@@omegaman7377 Square waves have higher order harmonics, technically infinitely high, compared with their fundamental frequency. A pure sine wave is ONLY the fundamental, so you have your sampling idea completely backwards. Sine waves require lower frequency sampling to recreate without any assumption about the input signal, because you only have to reproduce the fundamental.
I worked for BBCTV for 27 years as a sound operator and was there when digital recording reared its head. This video is most interesting and actually rather reassuring. Thank you!
How can we get the English version of "The Flying Dutchmen"
Thank you for uploading this! I saw Monty's video over 10 years ago, but just a few years ago I tried to find it again and couldn't. This video came up in my YT feed and something told me it might be what I was looking for.
*Insert MJ in Thriller eating popcorn gif*
Too bad GIFs aren't allowed on UA-cam! Thanks for watching, John.
@@AudioUniversity many things are not allowed here. Better go to concurrency.
@@AudioUniversity UA-cam does not allow a video of someone doing a finger stick test to check their blood sugar, the drop of blood might offend or scare someone...sigh.......
My best argument FOR the CD-standard is that it is not something they came up with and then proved to be good enough with math. The CD-standard was based on the proven math and was actually extremely stringent for its time. It's not more complicated than that with good electronics.
Things don't always work as well in the real world as they do on paper. Time travel has been proven mathematically but still not perfected.
@@acoustic61, well, that's true, buuuuut we HAVE empirically proven, as seen in this video, that the math holds for band-limited digital signals, and the same goes for the CD format.
@@TokeBoisen Music is more complex than some demo of a sine wave. I've listened to thousands of hi-res digital transfers and virtually every one sounds better than CD. I think it's easier to get better results with higher sample rates. Maybe because steep filters, which are imperfect, and other forms of processing can be used more sparingly. I see no reason not to use higher sample rares. Digtal storage is dirt cheap.
@@acoustic61 oversampling sounds better if any form of saturation is involved. So you're right that it won't matter for just a sine wave. Dan warrel has a good video on oversampling.
@@acoustic61, the sine wave is just the easiest example to demonstrate the concept. As is explained in the longer video from Monty, any complex waveform of a band-limited signal can be perfectly captured and replicated. If it deviates from the original waveform it MUST contain information above Nyquist and is therefore no longer a possible solution.
It HAS been documented and verified that phase-differences can impact the ability to capture perfectly if the chosen sampling-rate is exactly twice what you'd want to capture, but that is inconsequential for either CD or modern digital formats where the sampling-rate is much higher than twice the upper limit of human hearing.
Additionally, if you ever look at FFTs of hi-res transfers you'll most likely see that there is either no information above 20 kHz, or what is there is just noise. At best that means they just waste bandwidth, at worst it introduces IMD in the audible range. Any differences you are hearing are more likely to be due to a difference in mastering, an increase in gain, or just simply psychoacoustics.
*proceeds to change all settings on my audio drivers to 44100 CD and leaving them like that forever* It helps so much with eliminating higher processing needs that might end up in artifacts such as stuttering or crackling of the samples. Thank you! This has literally changed how I use studio equipment.
I learned this while studying computer science. It was one of the basics required for the bachelor. The same methods/theories are used for all kind of signal processing like WLAN or cell phone networks.
Thanks but I prefer analog WiFi. Reddit just has a better more real feel to it.
it's also used in television, radio, basically, if it has a signal, it likely at some point in the signal chain, uses the logic of the Nyquist sampling theorem in it's design.
is that why my cell phone sound so good?
@@oldolfmann8927 It's why the most common word in cellphone conversations is, "What?"
@@davidryder3374 my comment was sarcasm LOL. Digital does have it's place, but there are a lot of places I do not like it.
Thanks for mentioning at the end that 24bit and higher sample rate is important for production.
Indeed, for recording one doesn't need more than 44.1 or 48Khz, however in production when there are tens of filters and effects applied to a sound, distortion and noise is amplified if the resolution is insufficient.
I can't think of anything that would create either noise nor distortion if the sample rate at which the audio is recorded at 44.1k. Oversampling for non-linear processing exists and for linear processing it's not necessary (and for things like convolution a higher sample frequency makes everything eat up computational resources. My DAW supports 16xOS, applicable to either plugin chains or single plugins, which is plenty. No need to waste CPU power and thus performance on using higher sample rates for things that don't benefit fromthem...
@@simongunkel7457 They produce ultrasonic harmonics, it you don't have space enough on your bitrate to accomodate them, they "bounce" back to the sonic range and they can be heard as "artifacts". So you need space where they can propagate, and can be cut clean when converting back to 16 bits, because all the "trash" is out there beyond the 16 bits resoultion.
They will explain this on the next video for sure.
@@framegrace1You are confusing sampling rate with quantization (bit depth). The bounce back happens if you have non linear processing (like all distortion) and the sampling rate is too low. Quantization only affects the noise floor.
@@framegrace1 You are confusing bit depth and sampling rate here. There are good reasons to record at 24 bit depth, but that doesn't have to do with ultrasonic harmonics, but it allows for more headroom when recording and more tracks to be mixed (you lose about a bit for each quadrupling of tracks, so if your project has 64 tracks that will mean your output will have lost 3 bits compared to the recording. And you also lose a bit for each 6dB of headroom while tracking. When 16 bit was state of the art, you had to track really hot, which made clipping likely. With 24 bit, you don't have to track hot at all and won't clip and still have plenty of bit depth to allow you to go ham with multitracking). Aliasing, i.e. ultrasonic harmonics bouncing back from Nyquist has to do with the sampling rate. But as I mentioned you can use oversampling to deal with that. So my recorded audio at 48k goes to a non-linear plugin. It will alias. I enable 16x oversampling for that plugin, which will make the plugin see a 768k signal where the ultrasonics get to live up to 384k and then have another 360k before they would bounce below the 24k where they could end up in my 48k signal. That is then filtered out and the signal is converted back to 48k. But if I send my 48k signal to a linear plugin, it won't produce any harmonics and thus I don't need oversampling and will just run the 48k signal. Convolution is usually the most intense processing and it is linear, but the computational load scales with the sample rate. Note that my line of thinking here requires a DAW or plugins that oversample. I'm using reaper and per plugin oversampling has only been a feature for less than a year and not every non-linear plugin has internal oversampling (though plenty do). But even then tracking at higher sample rates rather than processing at higher sample rates didn't make much sense and now you can get granular with that. If you hit everything with 96k, you will slow down things that don't produce aliasing anyway and things that do will still have bigger issues than if they specifically got 768k signals to work with.
Exactly, processing in the digital domain will degrade the signal, unless you start out with a higher quality already. Just a simple +3dB filter already throws out one bit of dynamic range, so you better start out with some headroom.
Somehow audiophiles got wind of the fact that these audio formats exist and that it would be better to listen to this original mastering instead of a downsampled copy, which obviously makes zero difference, unless you can upgrade your ears to higher dynamic range and higher frequencies.
Although I'd still say that for typical audio mastering, only the bit depth will actually matter, unless you have some serious slowing down planned. Otherwise the high frequency components would remain outside the audible range either way. Maybe it might protect better against some quantization artifacts when filtering.
I was an avid amateur audio constructor ever since my teens in the 1950s, always striving for the best sound quality I could afford. It took less than a minute to convert me to digital forever the first time listened to a CD!
Thanks for this video.
I remember how the first CDs had a high end scratchy sound. It was because the master was mixed that way because they knew the transfer to vinyl would attenuate the high end. So when they made the first CDs they just grabbed the original master tape and sampled it to digital. And the digital faithfully reproduced that extra high end instead of reducing it like the vinyl processing naturally did. End result was digital got a bad rep. Later they remastered with digital in mind and the next release of the same album was much better. So I ended up taking some of my early CDs and tossing them and replacing with newer versions.
@@kthwkr Ah, the (in)famous RIAA equalisation! That chapter gratefully closed forever, along with tracking weights, lateral compensation, surface noise etc that went with expensive turntables and magnetic cartridges.
The current infatuation with vinyl is beyond me.
@@kthwkrwas it like a crt whine?
@@jq4t49f3 Add to that the infatuation with tube amplification. I was there when the conversion from tubes to transistors happened and losing the distortion and bias of tubes was a big step forward.
@@ClareHehe Think of it more like turning up the treble control all the way.
Glad you mentioned the issue with trying to play output that contains ultrasonic content may cause distortion. Many speakers will not have the frequency response to reproduce signals with ultrasound components as they were not designed to and they are very far from the frequencies they were optimised to reproduce best. This will usually be 20Hz-20kHz for a single driver speaker. The reason why no speaker can reproduce all frequencies equally well is that the strength of the electromagnet has to be tuned with the mass of the cone, the range of travel of the cone and the desired frequency response in mind. Play frequencies too low through it and the cone may move too far and physically clip out, play frequencies too high through it and the cone can't really respond quick enough for the smaller fluctuations in the signal producing noise in the mechanism that way. This is also why high end speakers often have multiple drivers fed through band pass filters, the big heavy driver that can push lots of air for that thumping bass is really not suited to producing tones in the higher range that need a more nimble cone or more powerful driver. The problem is putting in a driver that can move that monster through it's full range 15000 times a second and it will hit the limits of it's range of motion part way through the cycle with a 20Hz wave of the same amplitude. On the other end optimising it for a few hundred Hz means the cone doesn't have time to move far before the signal switches at high frequencies. So you can't have your cake and eat it and trying to demand a speaker to do everything will make your listening experience lousy.
It's actually the output filter that restores the original analog waveform. So the quality and steepness (or rather the lack of steepness, and here come oversampling and dithering into play) of the filter are essential. Maybe it should've been mentioned in the video.
Edit: Oh, I see Max Nielsen has already explained this in a much more professional manner. So it is with those who write before they read.
Agreed. A lot of people say that it's not a stair step, but the digital chain goes Dots from the PCM file -> Stair Step at the DAC chip -> Smoothed out using a filter. Source: Texas Instruments, maker of DACs
Many people have not grasped the entire chain and seize upon denouncing the stair step saying it does not exist. It might not exist at the speakers, but deep inside, it does exist. That some people fear the stair step at the speakers is perhaps reasonable in the case of an extremely cheap system that makes no attempt to filter and uses 8 bit, 11 KHz sampling which is barely adequate for voice and the sampling artifacts are clearly audible unless heavily filtered, in which case the heavy filtering is noticeable.
@@thomasmaughan4798 Audio University has given out so much misleading information that has led to online arguments that someone needs to do a debunking video. This video title is very misleading
@@thomasmaughan4798 And where do you buy 8 bit, 11khz discs?
Agreed.
As a member of the general public without any background in sound engineering who always heard of the stair-step wave thing and how digital sound is "worse" than analog because of it... Your explanation is SUPER on point. I was able to fully understand the concept. I give you kudos for being able to create content valueble for specialists and laymen alike. You're awesome! :)
Me too! But I had trouble understanding some points. For instance, I've never heard of the stair-step at all, so I had no idea what it even was, honestly.
Yeah, but he's wrong.
@@toomanyhobbies2011 elaborate
@@toomanyhobbies2011 well, he presented a long video with a lot of sound, reasonable argumenting for his point. You'll need more than "he's wrong" to be taken seriously. So please elaborate.
The people that tout "worse than analog" usually compare the most expensive analog solutions to the cheapest digital ones, in which case you're just comparing the relative quality of the equipment. But similar goes for the other side of the argument, if you compare a high-end digital solution to a bargain bin analog one the more expensive one will objectively be better, but again not because it's digital.
If you get something decent and keep the signal properly intact from start to finish both will be as good as the other, with some minor changes in the tuning depending on what all is in the chain at what point. I have both and enjoy using both depending on what I want to listen to at any given point.
Analog is a bit more prone to outside interference like ground loops or cables picking up noises from a nearby power line, which attracts purists that want to feel better about themselves using a harder medium, but digital is much more convenient and thus easier accessible to everyone.
Good music should be able to be enjoyed by everyone, not just elitists.
Higher bit depth though is used in the production process to be able to do things like adding two loud signals or other processing functions that would otherwise add quantization noise (ever see the bands in a dark part of a streamed movie?) or blow past the highest signal of a lower bit depth format. The signal is then requantized before mastering back down to 16 bits. The creators of the CD format really knew what they were doing, and had over a century of digital and signal processing research to call upon to come up with the 16 bit/44.1 kHz format.
Same thing for sample rate: They have a use in production. When you want to do non-linear filtering without the higher harmonics getting aliased back down, or because it helps to avoid filtering artefacts, or so you can slow down a sample without it turning really base-y. But those are only intermediate stages, needed to help the mathematics of transformation work. Once production is done, it's all turned back into something more suited to limited human hearing. 44.1KHz, or sometimes 48KHz.
@@vylbird8014 Exactly!
All true. High sample rates and bit depths are very important for production. But it's a complete waste for distribution. But it's not ALWAYS important for production either. Aliasing is not as common a problem as most people think. Not everyone is pitching and slowing recorded content down. And as for the higher dynamic range - it's very helpful to have that headroom in a recording, but again, is often not an issue. A lot of producers use it as a safety net, even though they rarely need it.
I'm more excited about the move to record in 32-bit float. That's not about dynamic range, but about never clipping unless the mic itself is overloaded, and never having to worry about gain until you are mixing. It's like the audio equivalent of shooting photos in raw.
@@mwdiers That would be really cool. Eliminate an electronic stage for gain adjustment in recording?
I thought the 44.1khz, 16 bit was due to Umatic tape restrictions?
Thanks for taking Monty´s great video to the "next generation". Used the original a lot in hifi-forums, not sure it helped, because many audiophiles share flatearther-vibes. But still one of the best explanations.
It is so nice to get a working knowledge of all these numbers and terms in my receiver manuals and software app settings. I am a slow learner butI am getting better and better sound to my ears (fidelity?) outputing digital optical from winfpws10 to a Denon receiver with 7.1 The quality of playback across apps in various formats like games or movies does vary and can be unpredictable but your explanation of termas and how they apply in practice are great. I was doinhg rresearch for clients DAC and found channel very helpful.
Fabulous. I have understood this although at less technical level until now. That testing method he shows is absolutely brilliant and his explanation is spot-on. Those first engineers at Phillips and Sony who created this standard and the Red Book, they knew what they were doing and set the bar at a very high level. This is a fantastic addition not only to your content library but for the general public.
Thanks, @Grand Rapids57! I agree - the bar has been set incredibly high. Unlike developments in video recreation, audio has really stood the test of time since the CD.
No wonder - although the result was wonderful ! Audiophiles were much more inclined to testing equipments… And, what’s more, the human ear is an outstanding tool, much better at discriminating than the eye : visual illusions are plenty, but aural illusions are just a few. And if you want to know the material of a wall, your eyes will easily mess up. Knock it with you fist and you’re ears will tell ! 🤓 Hearing is knowing, as exemplified by the fact that no one will make fun of a blind man, but a semi-deaf man is a funny character because he « doesn’t understand » and keeps making stupid mistakes… In French, « entendement » is what translates as « understanding », suggesting that « entendre » (I.e. to hear) is intimately linked to the knowledge of the outer world. Outstanding video, by the way. Deserves much more views ( sound included), even with the sloppy YT compression !
Philips wanted to stick to 14 bits in the beginning and only a short period before the market introduction the specification was changed to 16 bits under pressure of Sony. The first Philips CD players used 14 bit D/A converters and they implemented oversampling to end up with 16 bit accuracy.
Philips already had the design of their DAC’s ready for production so there was no time to create a new design so the clever Philips engineers came up with a 4x oversampling design using the 14 bit DAC to end up with 16 bit resolution.
Harder than Phillips stick with that before switching their stuff natively to 16-bit? Did it have a bad effect on their sound?
@@actionjksn they couldn't stick with 14 bits since the CD format was already determined to be 16 bit at that time. Using a 14 bit converter without oversampling would have caused degredation.
In the end it didn't cause any downsides in terms of sound quality. Philips was also on top of there game in that period, they produced some of the best DAC's in that period.
If you want to hear a good example of quantization noise, the Game Boy Advance has digital sound channels that are limited to 8 bits. Accordingly, many games sound like they have tape-like hiss, but only when sound is actually playing.
This is one of those videos I hope I come back to in ten years and completely understand. Like, I know there's some wildly smart and mind-blowing stuff being talked about and shown, but most of the video goes over my head. Still, it's fascinating to watch. Thank you so much for putting this content out for people to watch! I can't wait to fully understand this!
Higher sample rates are great if you are a sound designer and wish to do lots of time compression and expansion, particularly if you wish to get fewer digital artifacts. There are also times when you want the digital artifacts from time compression and expansion, so it’s all about that your desired effects are. Knowing when and how to use each audio sample rate and bit depth is a skill you get from experience. Great breakdown on the technical side for those not in audio production. Can’t wait to hear your thoughts on how to use each sample rate and bit depth when it comes to professional audio production. I wonder if you will get into more than just music production, but also pre and post production for TV/Film/Radio/Podcasts, Sound Design, Foley, Voice overs, and Voice Acting to name a few…
Mixing down to 44.1khz 16bit is perfectly fine and sometimes preferable, it’s only when you start messing with audio’s time and pitch do you really start to see immense utility in higher sample rates like 196. It’s great as a Sound Designer when you get to choose between different sample rates and bit depths, it adds so much flexibility when tweaking sound effects and voices. It can save lots of time and money if you have the ability to tweak sound effects and ADR VO to work and not have to record new sound effects, or especially bringing back in an expensive Voice Actor just because you are limited because of sample rate just how much you can manipulate a particular voice/sound effect…
Just my two cents…
Off course high sample rates matter when it comes to sound manipulation in samplers/DAWS/vst-i:s. But his video was only about sample playback for a fixed song with a wave that isn't manipulated. Even back in the 85 when the Fairlight CMI 3 was released, they knew that playing back 100 KhZ samples pitched down an octave would get you really cool sounding low tones that still retained "full quality" with less artifacts!
Oversampling is also used in many softsynths when they are doing "internal calculations".
So much disinfo here I don’t know where to start. Without realizing it, you are propagating the very silliness this entire video was made to debunk. I suggest you study dither until you actually understand it for starters. Then do some more reading to find out how top engineers actually do their digital recording. Sound Design is a somewhat low demand version of music, not the opposite. I’m happy for you to feel special about your chosen line of work, but it would behoove you to know what you are saying before you write these ridiculous screeds.
Some of the reasons to sample higher than 44.1 kHz was to make anti aliasing filter less steep to avoiding aliasing. This filter is in the frontend and analogue, so the filter is cheaper and gives less phase distortion. If the signal is oversampled many times you can take advantages to noise shape the spectrum (place the noise in the frequency range you will filter away).
Sigma delta converters do just that.
@SR-ml4dn In the original article, Monty does mention that *processing* at higher sampling and depth is perfectly sensible, the original article (which is linked in the description) was a response to "high definition audio". And the video here is a complement to that, clarifying misconceptions about digital audio & audio waveforms.
Front-end analogue filters are a problem because they can't be sharp enough without introducing distortion and noise. Instead we just sample at, say, quadruple the rate (176kHz) so the analogue aliasing filter can be much less sharp. Then we use a digital filter before down-sampling to 44kHz. Even digital can have trouble with the sharp cutoff between 20kHz of hearing and the 22kHz nyquist limit. Moving the nyquist limit up to 24kHz helps which is why 48kHz sampling is so common in newer standards, including digital video formats.
@@rogerphelps9939most/all silicon based sigma delta converters create idle tones and some cause noise floor modulation. This is why engineers like Bruno Putszeys designed discrete PWM conversion methods.
Came here to say this having a masters degree in vibrations for mechanical systems. Aliasing and side lobe distortion is always a major issue when recreating frequencies from sample data with FFTs and I see no reason why it would different here.
Key to this is limiting the bandwidth of the sampled signal. With proper low pass filtering, frequencies above Nyquist are eliminated and there will be no aliasing noise. However, oversampling can simplify the input filter complexity and the high frequency content (noise) can then be filtered digitally and downsampled to generate the output digital stream at 44.1K or 48K samples per second.
Internally audio production software uses 32 bit floating point to avoid clipping the 65536 values a 16 bit software synthesizer can generate. The final output is 16 bit though. 32 bit floating point is also faster for cpu calculations. In this video only the end results are matter-of-fact for human hearing.
There should be aliasing at 15 khz. Even if you generate the perfect waveform on the PC, there are clearly audible aliasing byproducts.
@@dtibor5903 2 times 15khz is 30 khz, but UA-cam cutoffs everything above 16 khz fo better recompression.
That's why it's 44.1K rather than 40K - need a bit of extra room because real-world filters are not perfect, you can't have a brick wall. Well, you can with oversampling and digital filters, but that wasn't around back when CD was introduced.
I have lots of different music formats: SACD, MP3, FLAC, DVD-Audio, Blu-Ray, and on and on. This video is in line with my observations. The original recording, regardless of the format, is much more important. But I still like high-density formats. I cannot tell the difference between ANY of these formats, EXCEPT when I crank up the volume. I can crank up the volume on my giant speakers maybe 5-10% louder (which I love) without distortion with high density formats (and I have plenty of the same recordings in multiple formats). But I always assumed it had more to do with the way the sound is processed by my specific electronics than by the actual bit rate. It varies by specific systems. This video implies that the noise reduction of higher bits may be the culprit, but given all the electronics issues involved, it's hard to pinpoint. Same with vinyl - I can get some unbelievably sweet sound from certain vinyl that just isn't the same from digital, but only with excellent (but rare) specific records, and only with crazy complicated setup and very specific components I've discovered via long-term trial and error. So again, it seems to have more to do with the specific setup than the actual music format. I love vinyl, but I laugh at the complexity of getting truly great sound with it - so many variables.
I'm very happy to see people start to explain this to the masses. For 2 decades I've been telling audiophiles (extremists?) that they rarely have less than 2-3 bits of noise on the analog part of a 16-bit sample and they don't believe me. Even explaining that 16 bits means 16 microvolts of resolution per volt and that it's easy to get more due to RF radiation around. Not to mention that some such people nowadays use class-D amplifiers and find them good while these ones generally provide less than 8 bits of resolution due to using an insufficient sampling frequency! Thanks for this video, really!
I knew this had to be about Monty as soon as I read the title. That video was seriously one of the best explanations of digital audio on the internet.
Thanks! It seems that we are reaching a point where younger generations are getting educated on these topics, and ruthlessly debunking all the old myths. I'm enjoying every moment of it. The same thing is happening in the world of electric guitar. Shout-outs to Jim Lill.
His videos are awesome, seriously. The guitar tone one blew my mind
It's hilarious to see guitarist's creating ridiculous theories to "debunk" Jim Lille's vids to justify buying their guitars for the tone wood
It's not an "old myth", this video is in error.
Exactly! Makes my day to see Jim Lill mentioned here too.
@@wizrom3046 how so, please explain.
So Good!! Thank you for your explanation and Monty's. The "zero-order hold" demonstration with the analogy of pixels really sealed the deal, and I thought "of course!". It is amazing how we can think deeply on a lot of things, but there are these basic assumptions usually imparted by inferences made from visuals cues that can distort the accuracy of said thinking. Think about how the Rutherford model of the atom (or really any depiction of atoms) informs thought about how atoms *must* work.
I became interested in electronics when I was a kid during the mid 1970s. I discovered high fidelity audio as a pre-teen during the late 1970s. At one time, my dream job was to become an EE and work for McIntosh. Then during the early '80s, I added computers to my list of passions. In college, I earned a degree in computer engineering, but I also took all of the electrical engineering coursework as well, which covered things like analog circuit design and analog/digital signal analysis/processing. My professional career has been in computer networking, operating systems, and embedded software development. But I never lost my love for (vintage) analog hifi. (It just took a bit of a hiatus when I was busy raising a family. Now it's back in full swing.)
From my educational background, I always understood that the "digital stair step" was a myth, but this video helped further elucidate that fact. Nice job.
I remember the origins of the CD and people dismissing digital as chopping the sound into little bits, that just has to sound awful! But of course most audiophiles are not that technically educated. Maybe just the ones that design and build the stuff. And their opinions are thus also tainted by wanting to sell you something.
"But of course most audiophiles are not that technically educated."
Whereas you are wise and smart.
When someone spends a lot of money on something, they'll do anything to convince themselves it was worth it. It's all that is!
@@thomasmaughan4798 Does one have to be all that wise or smart in order to be technically educated in a topic?
(In case there was any ambiguity: I don’t mean this as any kind of dig at people who are technically educated in the topic)
@@drdca8263 Some people are offended when their ideas are proven not to be possible.
I didn't really think CDs would sound good when I first heard about how they work, but I also didn't think of myself as some kind of superexpert on all topics so I just listened to a couple. I had friends who swore the quality wasn't equal to vinyl but I couldn't hear any difference other than the vinyl was noisier. So... I bought myself a fair number of CDs.
I remember watching that way back in the day. That original video was excellent, and it's a message that definitely bears repeating.
I've been looking that original video for ages since I saw it years ago. It clears up so many myths, especially he confusion between sample points (infinitely small) and the "fatbits" version audio software tends to show (though nowadays some actually show the equivalent of what the DAC will output, with proper curves).
Finally. Thank you. Honestly, anyone who thinks 96k is better than CD quality, I'd love to see them in a blind test. I am pretty sure no one could hear the difference. Nyquist rate etc. Also, anyone who COULD tell the difference would probably be under 21 and wouldn't be able to tell the difference at 30. Frequency range loss with age is very real and measurable. And most record producers, at least on a commercial level, are over 40, with many over 50 and a lot over 60. I doubt the ones over 50 could tell a good lossy format (say, 320k MP3, 44.1, 16bit, true stereo), from 96k WAV.
@@nicksterj Everyone's ear is different. If you cannot hear the difference between capacitors, high res is useless to you. ))
@@GBR9794 to hear above 20 Khz you have to be a dog.
@@brianhead814 it was a joke though
Good and informative, (old CS/EE engineer with audio interest here). Remember back at uni in the 90s we had labs doing exactly this, learning about different types of wave reconstruction from digital data in DA-converters (there is more to it than the video mentions but the conclusion is very valid).
One thing that is not mentioned with dynamic range is that there has been historically hard to have a standardized level for where to put the mean audio level in recording digitally. This is what is called the "loudness wars" where listeners tend to like music etc at higher volume. Also the reason commercials are usually at a way louder volume. This drove recording industry to record at higher and higher levels.
So a lot of CDs actually had a very high average volume meaning that it was less dynamic range to use. This made them sound inferior to what the CD-standard actually allowed. A lot of CDs were "badly" recorded so to speak. Ideally you would want the digital recording to be fit within the dynamic range in a way that maximizes the format.
The effect of reduced dynamic range is similar to how "nightmode" sounds.
This is one reason the classic vinylrecords can sound better than CDs. The format in itself is wildy inferior but they were mixed a lot better.
There is also another issue related to this and that is the intentional compression of dynamic range to sound better on devices that are not able to accurately reproduce sound.
There are some very interesting soundclips comparing speech and music with and without artificial dynamic compression. This is a very big area with lots of opionions (some mixers feel the compression is part of the style).
Personally I want my audio to have as much dynamic range as possible. To me it sounds a lot better, subjectively there is more texture in the music.
@@jim9930 Cool setup :) Yeah recording technique makes a huge difference. But with compression I’d did not mean compression as in bit/bandwidth reduction but as in compression of dynamic range. You hear it is a lot of modern recordings. If someone is singing it is the same volume regardless if they are belting or if they are whispering (a bit exaggerated but not too much). This is done in software usually for aesthetic reasons and for it it sound better on poor devices (cellphones, laptop, iPad etc etc).
Have listened to some horn systems and they have immense directivity and effectiveness. Tend to get very very big if you want to do good bass with them. Personally I have tinkered a bit with something called orthoaccoustic speakers that utilise room reflections more to create a more enveloping soundscape.
i love how a more or less old video explains is SO well and still people don't believe it and keep repeating confusing or flat out wrong info
Audiophile is just another word for superstitious, really. Never met one who had any technical knowledge at all.
What knowledge this boy has 😮 and the fastest growing channel of audio community.
I started following him when he had 77k subs now he is already on 200k+
I learn so much from him.
Thank you for sharing! ❤
Thanks for sticking with me for so long, DJ Suvy! Glad to help.
@@AudioUniversity No problem mate. Your knowledge is priceless. Keep up!
@@AudioUniversity Have you examined the myth of mp3 compression?
@@rafaelallenblock That's a video I'd like to see!
@@rafaelallenblock agree, let's talk MP3 quality.
What an amazingly informative video. Preconceptions about stair steps are out the window and the fact that quantitisation levels affect noise rather than the primary waveform was a real rite opener. Also love the fact you can put older analogue technology into a digital context like the compact cassette typically containing the same noise as a 5-6 bit sample!
Beware "just not so" stories on UA-cam.
"A cassette player is said to have a signal-to-noise ratio of 55 dB."
55 dB is 5.5 B, where Bel is the decimal logarithm of the power difference.
10^5.5 = 316227 power ratio; voltage ratio will be square root of that or 562.
9 bits is 512, so *you need 10 bits* to encompass the voltage signal strength dynamic range of a compact cassette tape.
As many parts of some kinds of music, classical in particular, ARE down in the noise, quantization errors at the noise level have noticeable impact on high frequency harmonics of some musical instruments. Noise is random;' musical signals are periodic. The Phase Linear autocorrelator worked by allowing specific high frequency bands in the noise when a lower frequency was detected greater than noise.
Clipping at the peaks of a signal is bad; but clipping at the lowest parts is also bad. You need enough "bits" to ensure you faithfully capture even the noise; because IN that noise is sometimes your signal and the signal can be recovered.
As a signal processing student and an audiophile - I thank you. Very nicely explained.
Students trying to flex😂
@@ClosestNearUtopia where in my comment do you see a flex? I'm expressing my appreciation for a good explanation, sounds like you're the one trying to flex here 🤨
Awesome. You sent me down the Rabbit hole with this one. I started reading up on this because I have been curious a long time. I don't claim to understand all of the theory, as a matter of fact I think most of it is beyond the capabilities of my brain, but the graphical presentations are very convincing indeed. Takeaways: bit-depth has no influence on reproducing the analog signal, only on the noise floor; Oversampling does not influence the shape of the analog signal reproduced; A block wave has artifacts because of cut-off frequencies; Dithering causes a nice even noise floor; Shaped Dithering causes a lower noise floor in mid range frequencies (where the ear is most sensitive) giving a better noise to signal ratio; Stepped curves are just a figment of the mind and misrepresentation of what actual happens. Thanks for this mind blowing video Kile.
Licensed professional engineer with undergrad degrees in engineering, physics, and mathematics and a masters degree in engineering AND a professional music producer here. You just got a subscribe. This was one of the best videos I've seen in awhile... and though I theoretically understood this based on DA design concepts, this video spelled it out with such clarity and reason, that I had to comment.
Great work!!
Just get rid of the lollipops. The output of DAC is latched, there are no lollipops and the staircase is real -- just microscopically tiny and passed through a filter to get rid of the 44.1 KHz sampling frequency.
@@thomasmaughan4798 When it's reconstructed to analog, it definitely isn't a stair step. Look at the output on a scope. Show me the stairs steps.
@@mc2engineeringprof " When it's reconstructed to analog, it definitely isn't a stair step."
It most certainly is! DAC can ONLY output DISCRETE voltages. You give it a 16 bit code, it spits out the corresponding voltage until it gets a new 16 bit word. Then the voltage jumps immediately (within a few nanoseconds) to the new voltage.
"Look at the output on a scope. Show me the stairs steps."
You CANNOT see 65536 steps on a scope with barely 8 bit vertical resolution! Noise will exceed the stair steps anyway.
Now, if you increase the vertical gain to zoom in on a *portion* of the sine wave, and you probe the output of the DAC itself, you will see staircasing. *That is how it works*
Some UA-cam videos compare oscilloscopes on this exact procedure. A digital scope creates staircases on *input* depending on its ADC. That's why you use an analog scope for this sort of thing; a digital scope itself introduces staircasing.
But what is the resolution of an analog scope? It is noise limited for one thing; 16 bits is a LOT of depth. A typical CRT for television has 512 lines of vertical resolution, it is impossible to see 65536 stair steps when the phosphor dot is already straddling hundreds of these steps.
@@thomasmaughan4798 You don't understand how D/A conversion works. Watch the video again.
@@thomasmaughan4798 you're describing audio DACs from the early 90s maybe, they haven't worked like that the past 25 years though. Modern audio DACs use upsampling and sigma-delta modulation to convert the PCM to very high sample rate (a few MHz) but only a few bits of resolution, so the unfiltered output may not look like the original signal but it also does not look like the kind of stairstep you're describing. More importantly, it's kind of irrelevant what the unfiltered output looks like, the external analog filter is essential to the proper operation of the DAC. The overall result is that it will properly reconstruct, to a high degree of accuracy, the (unique) bandlimited signal that passes through the sample points given to it, provided that this signal is within the passband of the DAC (which is at least 20 kHz when using a 44.1 kHz sample rate). The details of how exactly this is achieved is quite interesting but not particularly important to understanding digital audio as a concept.
Thanks for bringing this out Kyle. Excellent demonstration. I still use my DAW set to 32 bit float and 48 kHz, however, above CD quality isn't needed.
Look forward to the next video.
The problem with this simplistic experiment using a single frequency, continuous sine wave is that it completely eliminates any discussion of phase shift.
Go to a high end audio store and listen to some music recorded on a CD that is played back on a player with 4 and 8x oversampling. The difference will astound you.
Why? Oversampling permits the use of a lower order filter to reconstruct the analog waveform from rhe samples. What you don't hear discussed in the video is that filters (that reconstuct the signal) have varying phase shifts at different frequencies. Phase shift is time delay and that in turn makes the tones you hear move spatially in the audio you hear.
The closer to Nyquist the sampling rate the sharper the filter rolloff needed for reconstruction which in turn necessitates a higher order filter with, you guessed it, a much more distorted phase response.
So while 44 kHz audio meets the Nyquist criterion to recover the frequency content, it won't be sufficient to get all the phase info you need for crisp sounding stereo audio. Oversampling is critical for good audio.
Don't believe me? Like i said, go to a high end audio store and hear for yourself!
Oversampling in the DAC is digital interpolation between the recorded samples. It doesn't add any information. It just means you're doing some of the filtering digitally and the rest analog.
all modern audio ADCs and DACs use linear-phase FIR filters for anti-aliasing
I love the Digital Show & Tell with Monty! Every once in a while I re-watch them (parts 1 and 2) to freshen up my knowledge around the subject :)
Good discussion! I stand by my assertion that the best improvement in my stereo was the cheap CD player I bought back in the early 80s. Gone were the days of clicks and pops of vinal! It is interesting that vinal is making a comeback. I think it's the ritual of the turntable and cleaning that's the reason. I did double blind testing on very high-end equipment from the late 70s and early 80s and the CD always sounded better. The only thing close was a $5000 turntable/cartridge combo that was awesome! Sounded almost as good as a $130 CD player with a DDD disk.
My best takeaway is it's the music that's important and not the gear! You're far better off spending your money of good quality recordings than expensive equipment.
That said... Spend money on good speakers. Only the microphone (which you generally don't control) and your speakers should color your sound. Tube amps color sound in a nice way (even harmonics) but at the end of the day, it's still distortion! Great for guitar amps! And yes, I'm listening to this on an amp with a tube preamp because "tubes"! They are cool and glow! And have a VU meter! And that's through speakers of my own design. I'll switch back to a cleaner amp soon. But the VU meter is cool!!!
Thanks for the video!
That bit about pixels around 6:56 clarifies some confusions I always had about pixels when trying to contrast them with vector graphics
Arguably, that's bullshit too. Whatever you used to capture the original image DIDN'T have any sort of idealized pinhole-sized pixels - it had a surface area of some shape (actually rectangular most likely), and what it captured was the sum of all photons landing ANYWHERE WITHIN THAT AREA. Representing that as a square IS the sane way of saying "withing this specific area, this is the color we captured, and we have no finer-grained information concerning that area". And that remains true either for digital cameras or rolls of film - only there your "pixels" are the crystal grain of the film, of irregular shape...
Thanks Monty - a wonderfully presented video. Not just the actual presentation but, the actual content. How I wish I had seen this some 20 years ago during a debate with a so-called audiophile. He insisted that CDs could not reproduce the high harmonics needed to accurately represent an instrument, so he stuck o his vinyl collection. Although I do know that there are very high frequencies developed by, say, cymbals, I defy anyone to tell me if the 10th harmonic was missing!
Great video, very useful info and a succinct presentation - excellent!
I've been evangelizing Monty's video (and its accompanying article) since it came out. Glad to see someone else spreading the good word!
What a great video! I was aware of the technical inaccuracy of the stair-step, but I'd never seen someone show the analog signal converted back through an oscilloscope to show it that way. Very interesting.
Thank you so much for making this educational material available under Creative Commons, by the way.
All the sine waves I listen to sound better remastered in higher resolutions. I have a huge collection of sine waves, almost a CD for each frequency, and the higher resolution discs are fantastic. And the test equipment I play them on provides better imaging and soundstage than my Luxman amp and Wilson speakers.
I'm 43 years old and realised at a very early age that audiophiles are full of crap. Back in late 80s, we had LPs, 45s, and cassettes. That was it! And it was often common for you to record a tape with each new album, not only because it was more convenient to use on the go, but it was also a way to preserve that first play. Before your brother asked you to borrow it and get it all dusty and scratched.
So the first I listened to a cd, coming from records and cassettes, I was a young child. And you know what? Long live digital, it was the best thing ever happened in this world.
I was amazed how good it sounded, and I will never forget the cd either, invisible touch by the Genesis
How are audiophiles full of crap? You didn't explain it. You made that comment and then proceeded to talk about something totally different.
The fact that you think your shit sounds good while being cheap is nice. Good for you. But if I think it sounds like crap compared to my more expensive equipment, don't get all bent out of shape. You can both like CDs, and be an audiophile, or rather. A musicalphile.
Is there a lot of snake oil? Yeah, but that doesn't make it all snake oil. That's a logic fallacy.
I would hazard a guess that you haven’t heard your Invisible Touch played on a Linn Sondek LP12.
Funny when I heard the 8 bit hiss, the first thing that came to mind was the tape hiss I heard on my 80's tapes. And immediately he goes "well, that's like the tap hiss". lol
There is a benefit for using higher sampling rates, but it only applies when making the actual recording (I think this is what is alluded to at the end of the video - looking forward to the next video to see if I'm right!). Although humans can only hear up to 20 kHz, physical sound waves include frequencies higher than that. Those frequencies must be filtered out before sampling or else they will get aliased into the hearable range. But no filter is perfect and frequencies above 20 kHz will pass through the filter, though higher frequencies will be attenuated more than lower. So the solution is to sample at rates much higher than the filter's baseband bandwidth so that only the highly attenuated very high frequencies alias into the hearable range.
That's exactly the point. Ans also why amateur recording and reproduction of high bitrate without modification can actually sound worse than lower bitrates. Those are the ultrasonic harmonics that he talked about.
Well if you record a 20khz sound at 20khz, then every sample will get the wave at the same position, so the computer will assume it's just a straight line. So you need at least double that to make sure you can get both the peaks and troughs of the sound wave.
@@goldenfloof5469 Yes, however Nyquist is a double-edged sword. A 30kHz wave recorded at 20kHz will look like a 10kHz wave; that's why sampling high and then digitally filtering can produce nicer audio.
This is called "oversampling" and it's built into every modern analog-to-digital converter, even the cheap ones. There's no point in recording at a high sample rate and then downsampling; the converter does it for you.
@@allochthon I am an electronics design engineer; this is simply not true, sorry to say. That being said, you may be right with respect to half-decent converters which allow for multiple sample rates to be set. The problem is up-sampling and down-sampling with high fidelity uses non-trivial algorithms to implement in hardware. 48kHz -> 44.1kHz is an ugly conversion, for example, and a surprisingly common maximum for inexpensive electronics (though 96kHz is becoming more common). You can't just average data when down-sampling and get the same results as oversampling and then software processing.
You still need a filter behind a DAC to remove the stair-casing again but that's common sense for everyone who knows something about signal processing. I personally worked with a few microcontrollers in the past that were connected to some raw DAC chips. And they totally output stair-cases since everytime you change the value they just hold it until you change it again. You then need a filter afterwards which make the signal look like it should be.
A really important part, no mentioned in the video. The reason signals are presented as stair steps is because the sample value is held throughout the processing algorithm since sampling to outputting (which is synched via sample time and interrupts). So stair steps is a totally correct way of representing sample value in time domain, and they appears on DSP DAC output (they cannot be continuous by definition). DAC outputs in microcontrollers can be used used as debugging tools to observe DSP register values with oscilloscopes.
And you need 5-10x of the sampling rate to ensure the integrity and fidelity of the dynamic signal captured.
@@andytwgss What do you mean with that? You only need double the maximum frequency as explained in the video.
@@jangornjec6210audio class DACs have interpolation filter stages on the output, so don't appear like a sample and hold like a basic microcontroller utility dac will
@@NicosLeben sine waves for communication, yes; transient and dynamic signals, probably no. It’s the rise time that matters, not period. It’s a well known and common topic in high speed signal. You may refer to Tektronix’s documents about choosing oscilloscope.: The XYZ of Ocsilloscopes; Concept series: video/TV system measurement
Bandwidth directly affects the error margin in reproducing waveforms.
This was a great video. I am a retired electrical engineer. Most of my career was concerned with DSP (digital signal processing) of one form or another. I think that the reason that digital audio mystifies people is that you must have an understanding of some fairly advanced mathematics to understand why it works. Topics such as Nyquist sampling theory, anti-aliasing filters, spectral images, and Fourier transforms come to mind. Most people attempt to apply their common-sense analog thinking to it but that just doesn't work.
Very interesting, thank you. In my youth I had a hearing range (like you said), of 16Hz to about 19kHz, and an 'awareness' up to 25 kHz. I now have 67 year old ears and a very reduced audible range, so my 30 year-old record/CD/radio setup is quite adequate and still sounds as good as it did when new.
I would guess they use 24 bit high sample rate in music production for essentially the same reason movie producers and Photoshop artists use significantly higher resolution source material than the final product, because when the final rendering is done, any subtle errors in production will be ultimately diminished to the point they are barely if at all noticeable
Exactly, it's about preserving dynamic range of the finished product. It means you don't need to perfectly hit full scale range when you digitize the audio signal. You want to leave a bit of headroom when you feed any analog signal into an ADC so clipping does not occur. If you record at 16 bits and your resulting signal only uses 15 bits because you left yourself just a little headroom to avoid clipping, you can never get that bit back and your product can not be any better than 15 bits deep, or about 90 dB.
If you record at 24 bits and your signal only comes in 20 bits deep because you left some headroom that means you have 120 dB of dynamic range so you can hit your 16 bit/96 dB target.
Also, if you process everything at 16 bits then any added noise in the processing chain means your finished product cannot meet the 16 bit/96 dB dynamic range target of a CD.
I'm not sure exactly what advantage a sampling rate about 44.1 kHz would give you.
Yeah. I have a little pocket field recorder and I record at 24bit, mostly because then you don't have to care much about recording level - as long as it's there and it's not clipping you just adjust the volume in post
@@Fix_It_Again_Tony I'm not an engineer but a guess about one reason of probably multiple reasons why they might sample at 192k: When a studio records a track, each instrument/voice is recorded separately. Each track has overtones and such that have frequencies above the 22k limit. But when you mix them all back together, you end up with interference/resonance that results in some sound being audible to human hearing. A small, almost imperceptible part of the audio that you would hear if the band was all playing live in the same studio room
@@Sumanitu no, 192k is because nyquist thing, with analog filter you will never remove frequencies > 24khz and that will damage signal bellow 24khz during sampling. So low order filtering, 192k sample, then hi order (almost ideal) digital filtering to get rid of everything above 24khz and then resampling down to 48k
@@tkupilik the "nyquist thing" is literally what I'm talking about, bud. You dont want to lose frequencies above 22khz (1/2 the 44khz sample rate), even though they can't be heard by human ears. Thanks for re-explaining exactly what I was talking about...
What an eye opening video! Even with my engineering degree from back in 1976, my interest in building my own amplifiers, and my investment in high end headphones, this was a revelation. I have been wondering what to do about getting a better DAC than the one in my laptop, and you have convinced me that it's not worth going for better than CD standards.
I am also interested in math, and you just opened the door for me with the reveal that any deviation from the original wave requires frequencies higher than the sample rate. That is worth looking into as a mathematics exercise. I am thinking the DAC must use some form of interpolation between the points. The staircase is a brute force way, you could use linear interpolation, or better still quadratic. Then up pops Taylor series in my brain. Maybe you explain more in other videos, this is the first I have watched on your channel.
It doesn't use some interpolation. It uses a (low-pass) reconstruction filter (which can be as simple as an RC filter, i.e. just a resistor and capacitor, which filters out the stairsteps, after which you get the original waveform back.
Thank you, I now understand what I don’t like about my nephew’s band’s music. It’s as I thought, the sample frequency is to low. They’ve been compensating by over processing the signals.
I record at 44,1 to avoid downsampling, but use 24 bits to provide more headroom for effects plugins in each track. After mixing I export to 44,1/16.
Yes. There are several reasons to use higher bit depth and sample rate for audio production (recording, mixing, and mastering). That’s what the next video is all about. This video is just about audio playback.
@@AudioUniversity yes, for playback 44.1/16 is perfect. My CD collection is still growing and I really enjoy how good they sound considering how affordable a great sounding CD player is nowadays.
24bits does not give you more headroom. Headroom is loudness or volume above Odb(VU) or your normal operating level volume!
24bits give better sample accuracy or sound dynamics so does mean you are sending a better (more accurate signal to your effects)
@@AudioUniversity waiting for next video
@@rods6405 Depends on how you transform. You can easily have more headroom with 24bit and use it for expanded dynamic range, this is in fact one of the the primary uses of it before your compress it to 16bit at selected boosts.
If I mixed in 16bits, no one complained or even figured it out. I bet no one cares if the track is cool and has sufficient "musical nutritional value" in it xD People seem to care more about the content of the music than the quality or bit depth. Even if a top track is released on old tape, people celebrate it more than a track that doesn't touch them and that in 24bit. At least that's how I feel xD
That's a great point, 4N4LOG! The most important element is the music itself.
However, there are some reasons why you probably should still use 24-bit audio for recording and mixing (when possible). It's not a deal-breaker, but it definitely helps during the recording and mixing phase! Playback should be 16-bit.
And then there's lofi hip hop that trying to emulate average/below average tape sounds yet enjoyed by many
I remember reading that even the best audio systems in the world have the noise that equals merely 18 bits one(so no idea why 24 bits exists at all).
And unfortunately there are people who just don't believe Nyquist-Shannon sampling theorem no matter what, they somehow hear the difference...
Absolutely brillient. Very well explained and demostarted. I've seen quite a few videos debunking the need for HD audio on playback - everyone should watch this. Equally I've watched a video about high bit depth for recording and mixing, but at the end of the day it is normalised down to CD specification.
Useful. Thanks for posting. I've got lot of songwriting buddies just starting out. This will help them a lot.
Thank you so much! This was extremely informative. I've always wondered about this, since I couldn't hear the difference between "CD" quality vs higher bit and sample rates. I also enjoy all of your other videos! Kind regards!
It's a basics of Nyquist-Shannon sampling theorem. It's first year of Computer Science. One another example how "experts" say what "they think" instead of what's a scientific fact. Thank you for video!
Nice vid! Nothing I didn't already know, but you presented it well. As a composer and a real stickler for the details, I have been watching Dan Worral for some time. The way he takes apart VSTs is something else.
Excellent and informative video! Thank you. I am curious about two digital signals being mixed. During mixing, frequencies well beyond human hearing from two channels will beat against one another to make a new tone we can hear. A "beat frequency". How does sample rate affect the beat frequency produced? I've often wondered about that. Any ideas?
Fine work sir! My argument for all the vinyl lovers has always been db range. The maximum for LP was around 60db, not including noise floor. Then add in the pops and crackles after undergoing the RIAA standard. CD has been a great playback and preservation method (so far) compared to vinyl or tape.
playback yes, preservation not so much.. i used to work for an archives compagny and we prioritize CD over Vinyl and tapes, as the support is way durable if conserve in proper conditions. Average life time for a CD to start degrading is around 10 years if i remember correctly. It is way more for others medium
@@mambocountach I've been collecting CD's since the 80's and none of them have failed whether stored in a cool place or warm storage facility. Tape and vinyl don't do well in heat. I say this after retiring from the record industry after 30 plus years.
@@mambocountach I recall there were some CDs manufactured in the UK in the '80s that began to delaminate after 10 or so years.
Here in Australia a lot of early CDs were pressed by Disctronics in Melbourne, and to my knowledge none of those discs in my collection have degraded.
@@mambocountach My thinking is that there is no lifespan limit if the disc doesn't delaminate or get fungus(?) between the layers, a very rare phenomena. I have 40 year old Hi8 tapes that play like new, despite the experts claiming a much, much shorter lifespan. Supposedly, stored tightly wound the magnetic fields from one layer will imprint through the substrate to the next, mixing things up. Yeah, right.
And vinyl has a maximum bit depth of 12 to 13.
Fun fact: PVC molecules are big, and vinyl grooves are small. Together, these result in an equivalent dynamic resolution of 12 bit. Under ideal laboratory conditions.
But yeah, vinyl is "analog" as opposed to "digital"... that's just words and concepts. In physical reality, every 16 bit delta-sigma converter from the mid 90s is superior by orders of magnitude.
I had absolutely no idea I would enjoy being informed this much. Thank you
That is the exact explanation for my opinion/experience... and it is also clear to me that a higher bit rate makes sense in the course of a music production. Thanks very much.
It is a pity that the embedded video did not explain why the step function is not the end-result of digital analog conversion: namely that the signal is subject to subsequent thorough low-pass filtering. Without this information the nice harmonic output of a DAC is surely quite mysterious.
Yes, thank you
Yes, the missing reconstruction filter has been mentioned in the comments a few time. That it is there is indisputable. That it is not mentioned in the video in my opinion is most likely due to the video being a digital audio marketing video from back in the day.
Even the essential anti aliasing filter on the input is not mentioned probably for the same reason, to prevent the viewer from being horrified at having "filters" in the signal path.
Other than these simplifications it is a remarkably good demonstration and reflects the real life experience of digitised audio.
@@KallePihlajasaari As for me, I am delighted to have filters in my signal paths. They are beautiful results of human mathematical thinking.
Nyquist shows that the stairstep must be converted to Dirac delta conversion to simulate discrete time before lowpass filtering, so it's not the filter, it's the discrete-time approximation.
@@tomgroover1839 Huh? Anything can be filtered. The stair step is the default output of a DAC. Other digital filters may be used and these days often are but a low pass filter will remove higher frequency harmonic noise due to quantisation irrespective of what it looks like in the time or frequency domain. The point is that the lollipops are a representation for the mathematics on paper and are not used electrically. The electrical representation is the stair step until filtering takes place which typically is used.
RIGHT ON! 😎👍👍 Being old af, we learned all this when digital audio was first hitting the market.
It’s kind of a shame that you still have to walk people through this.
I think people are confused by the higher data rate that computers often need to PROCESS audio. This is a completely unrelated function.
Music is not a single sine wave of invariable pitch and volume. Thus you were fooled by this video. Sorry. In addition, bits don't equate to dB. Bits simply define a resolution range. You can use 16 bit to define 65536 different volume levels between 0dB and 1dB, or 65536 different volume levels between 0 and 1000dB. So here too you were fooled.
@@freecivweb4160 watch the video again and this time use your brain
I never even considered the idea that captured samples can only result in one possible output. That revelation is kinda of a game changer for me. Great video!
But that's the point in the video where suddenly there's no explanation where one is needed. :/ How can there only be one possible output? He shows someone drawing multiple possible outputs, then just says "turns out, nope, you can't." Why not? What if there had only been two or three samples? How could there be only one possible output then?
@@krisrhodes5180 if you draw a shape that is clearly wrong but hits the sample points, the you have tried to capture something that has way too high of a frequency for your sampling rate. Thats why sampling frequency needs to be double of the intended frequency range.
Okay I think I may be starting to see how this works. If I have just two samples, then the highest frequency this works for would be one (over the same time as the two samples). I can imagine that if I'm told "these two samples represent a sine wave at a frequency of one or lower," maybe there's a unique solution there. (Still feels like there should be a solution at every possible frequency one or lower but I can believe that feeling is wrong.)
@@krisrhodes5180 It's called the "Nyquist-Shannon sampling theorem", and I think if he went into the math in his video he would just lose the audience.
Great video! I want to clarify something about the Nyquist rate: it’s not necessarily that you need to sample at twice the highest frequency, but twice the bandwidth. If your signal of interest is, for instance, between 800 and 1200 Hz, then you can apply a bandpass filter that accepts that range and rejects what’s outside of it. With such a band-limited signal, you don’t need to sample at 2400 Hz, but 800 Hz! Essentially this technique exploits aliasing to reproduce high-frequency signals.
Source: I work with RF professionally
Great post! I wondered when someone would correct the faulty definition of the Nyquist theorem. I have also sub-sampled many signals in my days. Also, in the video it was stated that the stair step view is never there. You do get a stair step when you have no DAC anti-aliasing filter.
This is not entirely correct. By this logic a signal between 1GHz and 1GHz+400Hz would only need to be sampled at 800Hz. You're missing the downcoversion step -- mixing with the carrier and low-passing.
@@dmitryjoy Actually you can, without down sampling, as long as the input stage in the ADC has a sufficiently large full power bandwidth. Now you have to bandpass it to only let through your band of interest first though. Sampling at 800Hz with a 400Hz band of interest would be an impossible filter to make since it has to be a "brick" filter though. You also have to be smart about what sampling frequency that you pick. I have done sub-sampling implementations in a professional setting, with RF signals. The linked video explains it. ua-cam.com/video/ryJPVHrj0rE/v-deo.html
So true, however with CD Audio the dead bands below 20Hz is not worth mentioning as a credible saving.
In early A-Law and u-Law phone service codecs they used 8ksample with input filters of 200-2800Hz giving just 2600Hz bandwidth that in theory could have been achieved with 5.2ksamples with perfect fairy dust input filters but simply not worth the trouble for the savings.
The input circuitry on VERY high bandwidth oscilloscopes makes use of similar techniques elevating them to near magical levels with parallel sampling of delayed signals as required to compensate for the inadequate sampling rate but still maintaining the input bandwidth. So a 2 GHz input might use 4 x 1GSample ADCs or 8 x 500MSample ADCs plus a lot of DSP magic.
As a hobbyist, this video just made so many things make sense that id not understood before. Thanks!
A long time ago I wrote a program to reduce the bit depth and sample rate of audio recordings so my students could hear what difference it made. One of them had just bought a CD of Black Sabbath's Greatest Hits so we gave it a try with my program. However, Ozzy Osbourne's vocals sounded exactly the same unless the bit depth was reduced to 4...
The sample rate vs frequency experiment for a steady state tone was a quite convincing experiment. Sorry to say that in the real world of transient sounds, like music, it does not hold up the same. Coming from a Digital TV background I recall a demonstration at an Ampex class to demonstrate a problem called aliasing that happens when you have less than full quadrature sampling, in analog TV for Color information at 3.578545 Mhz. The normal sample rate used for Time Base Correctors and Frame Synchronizers is 14.31818 Mhz or 4 times the color frequency! This is to properly reproduce all phases of the original color signal at proper amplitude. In the demonstration the sample rate was reduced to 3x color, or 10.738635Mhz. At that sample rate there was very clear degradation in the color signal. There was aliasing causing amplitude and phase errors to appear, but for a consumer reference, it was still better than the average VHS home recording! When the sample rate was reduced to 2x color, or 7.15909 Mhz, only colors that phase agreed with the phase of the samples were correct with the amplitude of anything else was reduced according to the phase difference up to 90 degrees where there was no response. There were, however responses where colors changed as for a few cycles the frequency of the transition moves above or below the exact color frequency. These transitions, however, did not achieve the same amplitude traces as the original signal!
So.... with this in mind, how does the steady state tone demonstration come up looking so good in this demonstration? Why is it not full of aliasing products??? Simply put, it is the D-to-A post filtering that tends to ring to fill in nicely for steady state signals! By the way, the stair-step representation is exactly what comes out of the digital to analog section ahead of the out put filtering. The sampled values are loaded and latched as steady state values at this pre-filter point in the process.The output analog filter smooths this out.
When Digital audio was new the initial response from Recording Engineers and Musicians alike was WOW, THAT'S GREAT! It was then a leap in quality compared to Ampex 456 or the 3M equivalent analog tapes. They were really good, but always added a sort of what I call a "metalic sound" when comparing live to repro during a live recording. Digital was a lot better as the samples were very accurate in the majority of the audio range and as pointed out extremely quiet, lots of dynamic range. After a while, though, many Engineers and Musicians started noticing a sort of strange new component that most said sounded harsh! Believe it or not, that is the aliasing caused by less than full quadrature sampling!
As an example, one of many, I was recently approached by a Flute player who was frustrated by the sound of her recordings! They were done with a Neuman large diaphram U-87 type mic that has been a standard for decades. The recording was clean and clear except for some high frequency "trash". As I have seen many times in the past, a spectrum analysis clearly revealed that the digital signal was "growing" aliasing products from 11.025Khz (quadrature for 44.1 sample rate) and above as louder, higher pitched notes were played! I told her to have her engineer apply a simple lo-pass filter to the recording starting around 12 Khz and like magic the aliasing products were no longer a problem.
As for analog never having the dynamic range of digital, I am sad to say that both Ampex (Quantigy) and 3M came out with what they called "HIGH LEVEL TAPE" to try to recover from the digital revolution! They recommended setting your 0db record reference at 9 db above the standard 0db reference! I did this on my old Scully 280 and proceeded to do a run with a Sound Technology distortion analyzer. When the analyzer reached 27db above the elevated zero the distortion abruptly rose from fractions to 2.4%! I think that was the electronics flat topping, not the tape saturating! The headroom was great and when switching from direct to repro it was near impossible to hear any difference! Twenty years or so ago I recorded a live concert with my 1" 16 track using DBX and the Ampex 499 high level tape and the result was stunning! The recording was extremely quiet and dynamic sounding, but having said that, an XR-18 to a laptop with Pro Tools is a lot easier and sounds just as good!
One last thing, don't ever tell me that you have a "loss-less" audio compression thing! Like anything that compresses, it may be real good but not perfect! There is loss, but it may be hidden well!
I like most of what you said here, but as a software developer who studied computer science I feel I need to explain what is meant by "lossless" compression. A compression algorithm (for a digital file, not what an audio engineer means when they say compression) is considered lossless if the output of a compression-decompression cycle is identical down to the last bit. There are many lossless compression algorithms, such as the one used in the typical "Zip" file. And they are truly, in every sense "lossless". The same cannot be said of compression such as that used for typical MP3 files, which are lossy, as the result of a compression-decompression cycle is NOT the same as what was input. By saying a file is lossless does not mean it perfectly reproduces some live audio, it just means that whatever master digital file the audio engineer created for distribution can be delivered to you in a smaller size (saving network bandwidth and storage space) but the player will decompress it to be identical to what he created during playback.
There are advantages to 16 bit 192khz sample rates, That is in music production. This is because If you want to time distort an audio track or have it slow down (vinyl effects) It will sound better when it's running at a quarter of the speed because it won't sound so compressed and low frequency. Because all of the ultrasonic frequencies are also being brought down to within audible range so, of course after you do all of your editing, the final production can be 44.1 of course, but for production reasons, it's the equivalent of using vector or extremely extremely high resolution logos, because depending on where you move it and scale it and make it really big or small, it might affect it more or less.
I just felt a great disturbance in the vinyl market. As if millions of hipsters cried out that record players are better and were suddenly silenced
I buy vinys as fancy containers for mp3 downloads. They are nice display pieces.
I still like the sound of a good well preserved record. its the coloration which seems to have that ear tickling effect, not because its more accurate.
Vinyls still sound great, but it's more because of how they are mastered different from CD than anything to do with it being analogue. Can't have a vinal with so much range that it jumps your needle off the groove and all that.
@@lorestraat8920 Part of what makes valves so appealing to the ears is the harmonic distortion they produce. Its a pleasant effect. And vinyl has a similar effect provided the mastering is done well.
But where i truly thought where the debate would hold water, was when recording live instraments to my Teac Simulsync with ampex 456 on 15IPS. Vs recording them to a really good ADC into a computer. The ADC i have is Universal Audio equipped with all Burr Brown opamps and other quality parts. About as good as ADC can get. Using the same mic and mic pre, the Teac sounded better to my ears every time. Its like it preserved a certain warmth and depth that the UA ADC couldn't, and I even tried other recording interfaces as well.
Now having the UA ADC in a 64 bit floating point and high sampling did manage to match the Teac in that respect... But Otherwise I am unsure why the analog equipment was seeming to preserve the sound in a more pleasing way.
Its why I have tended to keep my recording in the analog domain while it makes sense to do so. Provided it is preserved without degradation from generational copying.
If CD really was as good as these people say then it would have killed off vinyl long ago. The problem for CD is that most musical instruments emit ultrasonic frequencies above 20khz. The CD just cannot reproduce those frequencies, but vinyl can. Also, these frequencies have been shown to have a positive effect on humans. But they don't want to talk about that because they are not particularly smart engineers. But ask the engineers who work for Sony and you will get a different perspective because they actually use their brains.
I'm not technically "smart" enough to actually understand all of the crazy numbers and math that go into actually making, recording, and mastering digital audio, all I know about it is what sounds truly good to my ears, (and what doesn't), and anything that is at LEAST "CD standard" sound quality OR higher, (as in TRUE "Hi-Res", etc.), sounds PERFECTLY FINE in EVERY way to my ears, and I can still hear pretty darn good at my age, and I always have been able to hear extremely well ever since I was a kid. (Up to 27.5KHz as a kid and teen, and still up to 17.5KHz now, at 50+ years of age)
And, ultimately, for about 99% or so of ALL music enjoyment, simply because of "where the actual music is", (as in what frequency bandwidth is needed to capture most of everything out there that we can actually hear), all that anyone really NEEDS to be able to hear for pretty much FULL musical enjoyment, (get out of your overly nerdy and analytically critical "head" and instead just listen more with your actual ears AND "heart" in order to be able to actually ENJOY your music, rather than just always "analyzing" it!), is really "only" up to about 16 KHz or so, so all of the people saying that they can "definitely" hear a "significant" difference between (and "improvement" in) "192/24" vs. "44.1/16" digital recordings, (or whatever other similar number comparison you want to make), ESPECIALLY if they're already relatively "older" and can ONLY actually HEAR frequencies UP TO anything BELOW 20KHz, are really only just arguing for the sake of arguing with someone about it, meanwhile TOTALLY forgetting to just forget all of the "numbers" and simply sit back, relax, and actually listen to AND enjoy the MUSIC itself, instead of always just focusing on the "specs" of it... On WHATEVER format or resolution that it's being recorded and/or played back on, especially if (at least as long as) it's at least as good as the basic "CD standard" format or anything higher than that...
Quit totally "nitpicking" your way out of actually paying attention to the music itself AND actually being able to fully enjoy it in the first place! That's the one thing that I see so many "audiophiles" doing nowadays, because it's really all just a "numbers game" with technology nowadays anyway... Pretty much ALL of which is better than most human's abilities to even hear or distinguish any significant difference between in the first place, so if you fall TOO much down into the endless "rabbit hole" of "high end audio", then you can even get to the point where you simply "CAN'T EVER be satisfied" anymore, because you're always chasing after the "latest and greatest" or "best of the best" newest technology, but yet you can't just simply sit your @$$ down in the "sweet spot" and just totally FORGET about all of the "technical jargon", specs, AND all of the "audiophile snake oil" and utter BS long enough to actually let the MUSIC or song itself bring true Joy and Happiness into your life, regardless of how "High-Res" it actually may be, (or maybe isn't)... Who really cares about that part of it anyway, it doesn't even really matter beyond it being at least as good as or better than YOU can actually physically hear, so just sit back, relax, take a deep breath, and enjoy the music itself for the beautiful track or musical piece or song or whatever that it IS! Quit worrying yourself so much about all of the "technical specifications" of it as long as it sounds really good to your own ears!...
The ultimate "format" and/or "resolution" of the music doesn't matter NEARLY as much as the actual music or song itself does, ESPECIALLY once it is at least recorded and played back at anything equal to or above "CD standard" audio quality! Always remember that much at least, and then you will ultimately be MUCH happier throughout your life, especially if you consider yourself to be at ALL an "audiophile"! (IMO, a TRUE audiophile knows how to thoroughly enjoy and appreciate their music for what it is, and they can actually let the music bring them joy and happiness, WITHOUT worrying obsessively about things like "bit and/or sample rates", etc! (Just for one example)
Monty did such a great service. That video helped me out immensely same as it does everyone who really watches it with an open mind.
When I saw the thumbnail, I was like "yup, that's what the LPF slightly below the Nyquist frequency is for, don't people know?"
And then it hit me. No, people don't know. Great video, thanks!
Apparently some people won't accept the Nyquist result no matter how much evidence you show them.. It seems to be almost a religious conviction. The same people probably have very little ides of what a Fourier transform is. I guess the only way to (maybe) convince them is to put them through a double-blind test situation in which they have to prove that their 'magic ears' can discern the difference. This has been done numerous times, I'm sure but I don't have any cites to hand.... do you know of any good links?
“Almost” religious? I would say it’s decidedly so! :p
And here I am at 71 years old and I can barely stand the 25kHz scream of the dentist's scaler. Those things are LOUD if you can hear them. As I age I'm losing my high-frequency hearing, but I can still hear these supposedly quiet machinex. One of my sons has the same range of hearing, and can't have the scalers used on him. Silent Dog Whistle? No such thing as fas as I'm concerned.
As for digital music? The problem I experience is the lack of dynamics. MP3 is the worst thing that ever happened to music.
I wish i can hear 20k again
Maybe you just hear the subharmonics when you are at the dentist. As to MP3, read the research by Plenge, Kürer, Wilkins, who came up with the mathematical model of the ear in the 70s, i.e., they analyzed what the human ear can hear and what not. The people at Fraunhofer then took their results and used them to create the MP3 standard...
This was very interesting, thank you! And here I am purchasing “high-res” music files off the internet and digitizing my vinyl at 24 bit/96kH, while every time I put on a good old CD I marvel at its audio quality. This has convinced me to no longer bother with ultimate audio resolution.
A HUGE thank you! Excellent presentation! The Philips/Sony engineers that developed the CD format were brilliant!
I was in college getting my engineering degree while Philips/Sony was developing the CD format so most of this information is NOT new to me, but what was truly mind blowing back then was storage density. My buddy and I were going through the College of Engineering junk pile and found two MASSIVE hard drives, and having just heard what the CD format was going to be, we started crunching numbers to use those junk hard drives and other instrumentation to kludge up an A/D and make our own CD format files from taped live recordings. Well, the whole effort ran out of steam when we figured out that the MASSIVE hard drives could hold at most eight seconds of music. Eight seconds.
Oh, and I'm still using the Philips CD player with TDA1541DACs that I bought in 1988.
Fantastic! I could never discern any difference in sound quality between Redbook and SACD, or higher resolutions.
I can, however, hear compression and its effects on badly engineered recordings, especially newer recordings.
Sure, 192khz (or higher) is a great way to record, since you always want to start at the highest resolutions to downsample later
for assorted formats.
Thanks. This answered so many questions about sampling that I was afraid to ask.
We figured out digital audio decades ago. Sure, improvements to things like CD players were made after (though often, those were actually meant to save cost to the manufacturer - despite what the marketing said), but the CD standard itself is plenty. It was back then, and it still is today.
This, and one other equation, was the only thing in my four-year electrical engineering degree I'd never heard of anywhere else. Thanks for making a video of it. When my professor proved in calculus that a perfect reproduction of a continuous wave exists in a jagged-looking digital stream, I gasped "what!" and the rest of my class looked disappointingly unmoved by the revelation.
It’s the magic of the Fourier transform.
How does this work? What if instead of fifty samples clearly visually suggesting a sine wave, I just have two samples? Is there somehow a way to reconstruct the original even then?? I don't know calculus so I know this may be a hard question to answer.
@@krisrhodes5180 Trying not to be too jargoney and also hoping I remember it correctly... "the Fourier transform of a rectangular pulse" can be applied to the concept of digital sample playback if each "stairstep" is transformed individually... the sinc functions overlap and the resulting waveform looks more and more like the original as you add all the harmonics of the stair steps.
He used ancient overhead transparencies to demonstrate it but I'm sure it could be done in Excel or MATLAB.
With filtering you can certainly approach the original waveform if it's a relatively simple combination of sine waves.
Thank you so much for making a video about this topic. It was high time, that someone with the proper reach does a video and debunks these myths. We really need videos like this! It’s just a shame, that nobody of the reviewers, who keep „selling“ a high sampling rate or bit depth on audio interfaces, will never really care to watch this video.
Problem is they CREATED far more false myths than they debunked. The only "debunked myth" was using the stair-step as an [over]simplified way to visualize how limited sampling rates always lose data. The actual mathematics for how sampling loses data is more complex. It comes down to using mathematical guesstimates in the reconstruction filters during the D-to-A playback. But didn't they just disprove that? No. Real music isn't the simplest wave theoretically possible--the 1kHz sine wave. Absolutely No. Real music is a true ocean of multiple waves and overtones with ever-changing apex and trough points which are escaping the sampling rate and have to be "guessed" by the reconstruction filter algorithms. These algorithms create all manner of "false local minima" and "false local maxima" -- basically you might call them hallucinated soundwaves artificially interjected into the playback.
Reconstruction algorithms therefore introduce false and lossy artifacts during the D-to-A playback. What Nyquist can do for a single invariable sine wave is true science, but only an idiot mistakes that for variable waves some of which aren't sinusoidal.
Moronic and utterly sophomoric and just-plain-dumb-as-f\/ck is the ignorant ineptitude of using a 1kHz sine wave of unchanging volume, as a proxy example for recorded music. Real music presents mind-numbingly difficult mathematical and engineering challenges which lower the quality of all digital playback at ANY AND ALL sample rates currently used by humankind. Want proof? Until you can hear digitally recorded cymbals sound like real metallic rainbow shimmers rippling through a 3D holographic room--which you can't--you have LOSSY LOW FIDELITY. 50 years from now they'll laugh at the arrogance of those saying we can't surpass 16/44k1. At 44k Hz sampling, cymbals sound like some someone pronouncing "ch ch ch" dubbed over the sound of shredding paper.
Very interesting video - lots of work went into it... thanks! I have one question I hope you can answer: I know what equipment you're using in the video such as the scopes, signal generator, and spectrum analyzer, but can you tell me what the device that looks like a tablet, running software with a blue background is and what the software it runs is ?
Well done. I should point out that higher sampling rates are better for certain use cases, such as playing back the sample at slower speeds. For example, if I shoot a video at 240 frames per second then play it back at regular speed (30 frames per second). Slow motion playback is far smoother than if I recorded it at 30 fps. The same is true of audio. So, if I record audio at 96k to sync with a high speed video, it will be smoother when I play it back at a slower speed.
Great tutorial, presented so well it cannot be misunderstood. A *little* knowledge is a dangerous thing. The right knowledge is liberating.
I believe the 44.1 kHz audio used by audio CDs, Was really because at the time with Sony PCM, that was the maximum that could be recorded as data on U-matic videotape for PAL and NTSC standards. And later on, Betamax was often used for digital recording distributions to the CD pressing plants. This was before DAT became the norm in the recording industry.
Ahh penny dropped moment. I took a tour of a CD pressing plant in the 90's when it was all quite new, and we were shown the process from start to finish. When the engineer showed us the source material as a videotape we screwed our faces up, I guess thinking it arrived in analogue form. As an aside, they had to destroy a large proportion of the pressed CD's as they had spots in them, usually black. They played fine, but consumers would think them faulty and return them. Another thing they showed us, and gave us a sample of which I still have today, was the ability to press a hologram into the disk instead of printing the tracklist or whatever. Looks amazing, but I have never seen it used on a CD release.
Thanks, the part about the digital sampling process makes total sense to me! I‘m not quite convinced that 24bit/192khz (or Sony‘s 1bit/2.8Mhz) doesn’t have any benefits in playback at all - but then again, I’m also not convinced any benefits can indeed be heard, either.
The difference between playback and production that you mention in the end reminds me of JPEG vs RAW in photos: JPEG is somewhat good enough for playback, but RAW is highly beneficial when you want to edit the stuff the sensor gives you because you need enough headroom. Still, even just for playback, 10bit color depth is better than JPEG’s 8bit …
I was so sure the fast fourier transform was going to be mentioned but I was dead wrong, super interesting video!
As a format, CDs offer twice as much dynamic range as even vinyl, but it's a shame a lot of music made today is just not mastered using any of it.
The dynamic range of vinyl was at best about 70 dB and CD is 96 dB or at the most 120 dB with dither, but yes, the last 40 years has been ruined by the "Loudness Wars". So sad. I haven't bought any music in the last 20 years.
The same could be said back in the day of analog recording. Hi-fidelity is usually saved for classical, with most pop and rock being recorded with compression play-back in mind, ie, AM/FM radio.
@@garymiles484 you're missing out, tons of less well known artists are producing beautifully mixed and mastered music. tired of hearing old people complain about music these days lol you all just aren't looking and listening hard enough
@@DAVID-io9nj It was necessary to use compression to make it audible in a car over the racket made by the engine.
@@EcoCentrist I am an old fart who started listening to music in the late 60's. Just because the recording process is great does not mean the music is great. Lucky for you to find new music you like. I had given up on new music until I came across rock from Japan!