That is, in fact, the technical term for it. It's under Clip > Transform > Un-upside-down.

5:01, I presume?

Oh wow, I didn't even notice. I just thought it was part of the presentation =D

You have to cater the antipodes as well.

In VideoPad, it's called re-rightsideupify

Yes that's right. There are 3 ways the brain separates the position of sounds with the help of our 2 ears. 1. Difference in volume 2. Difference in time 3. Difference in timbre The first point is pretty self explanatory. The second point relies on the distance between our ears and the fact that there will be a time difference left to right (or vice versa) between any sounds that are to the left/right of you. If two similar sounds arrive at your ears within 50ms of each other, then your brain uses this difference and interprets it as a position, favouring the side that received the sound first. Any longer and you will hear two separate sounds. The third relies on the fact that if a sound directly hits one ear, then your head will "shield" the other ear from the higher frequencies in the sound. This is because higher frequencies are more directional than low frequencies and are less likely to "bend" around your head. The side with more high frequency content is interpreted by your brain as being more likely to have heard the sound first.

C. H. The biology and high level concepts are fascinating and I can comprehend them pretty well. The physics, though, that's another story. The way multiple sound waves combine into a single two dimensional signal that still works for us as they're still distinct is the hard part. Harmonics, different sounds interacting by being amplified or cancelled out, etc. All that physical stuff that happens in between the 100 piece orchestra and the microphone turning their combined work into single series of numbers (or two for stereo :p)

I believe it's as simple as summing them. In the digital recording domain you've got a hard top limit to go by as well, so if you're using a basic DAW you're averaging the values and then amplifying them (or not) depending on how mush headroom you have. Newer DAWs with 64-bit internal precision mean that the summing can just happen without the master fader ever clipping.

This is also why you can have lossy coding like MP3 and it still sounds fine.

Just like a fourier series, it is indeed just as simple as summing them.

assuming that voltage is torque and current is rotation speed, I like to think to the capacitor as the spring and the inductor as the flywheel. In fact, the equations governing them are pretty much the same

Gianluca Ghettini this is like a perfect analogy!

Wouldn't that be inductors? Or am I reading it wrong?

TiagoTiago yes, my bad. Updated!

Gianluca Ghettini Ok, I get a flywheel resisting change in RPM. I get torque being Voltage. What is this spring? And how is it resisting change in torque? Is it between the torque source and the drive shaft? Is it line with the drive shaft and stores torque as potential? Also, is this instead of my analogy of a weight on a spring? It probably would have better maths because I guess it's polar maths? Amd when you watch the voltage try to track the digital input you're watching the flywheel edge on, rotated 90 through the vertical?

You can't worry about it. You don't have a choice between some oscillations and none unless you're going to invent a new technology. He pointed out how both analog and digital are exactly equally faulty in this sense.

You're taking my comment _waaay_ too seriously. I merely found the phase's use in thia video funny.

what timing?

“So the key here is: Don’t worry about it!”

@@littlegandhi1199 It's not faulty. That's just how analog sounds happen in the real world. It's that if you could project a perfect square wave, your ears will still have those ocilations. Because your eardrum can not physically teleport from the top to the bottom of the canal.

While it can't be questioned, about about the effects of dithering.

Some people can hear above 20khz though... Solution? 48khz instead of 44.1

Dalton what’s funny is that it’s possible that what the ‘audiophools’ are referring to a ‘more perfect’ of vinyl sounds is actually the imperfections that limits the analog signal: physical defects.

@@pilotavery Just going up to 48khz helps a lot, CDs 44khz make everything over 15khz sould more and more like noise where 20khz sound 100% like noise if it is more complicated then a pure sine wave...

@@a64738 Oversampling does in fact change the signal to noise ratio, I like a crisp high end but I question the "musical" application of frequencies over 20KHz...

Yes, that Monty. :)

Plus a hug for those royalty free codecs.

Monty also has a good video that explains A/D and D/A conversion. kzfaq.info/get/bejne/ma-BbKyJuLvepYE.html

This is hard to believe. You worked them out by hand all the time without knowing the basic idea of what it is? In my physics bachelors, I didn't use them "all the time." I used Fourier series in probably one or two classes.

+mohomed1208 i hear it's reached a very high level of development, its THE FUTURE

It's as bad as Dr. Who's timey wimey stuff.

All hands on deck! Swirly-thing alert!

is it timey wimey?

Yes, but the sufficient condition given isn't quite right. A sample rate exactly twice the highest frequency suppresses that frequency whole or in part, depending on the phase, and unless the samples fall right on peaks.

This whole time I've been wondering what happens if the samplerate falls out of phase...

A series of samples going high-mid-low-mid-high describes a frequency of a quarter of the sampling rate. A series going high-high-low-low-high-high also describes the same number of samples per cycle but at a different phase. If the sampling rate is an integer multiple of the frequency described, the exact pattern of samples depends on relative phase.

The work of the low pass filter is producing analog integration of signals... seems to me.

Yeah, too bad he got it wrong.

Agreed, everything he does is gold. I'd like to see a Technology Connections video on peanut butter just to see what he does with it.

I suggested the CD.... EFM, hilman diffie encoding, tracking, focus etc.... fits right in with your DAC suggestion.

That's pretty much this video. If your DAC is poor then reproduction won't be in such a nice waves as were shown here and there will be more artifacts (inaccuracies).

@fred I sure do, it's just a little more developed than a 10 year old's.

You see one in action if you use the falstad circuit simulator, click on the "Circuit" menu -> "Analog/Digital" and load up a DAC like the R-2R Ladder DAC. www.falstad.com/circuit/circuitjs.html

Agreed. That was the keystone that tied it all together for me too. Man this guy is awesome!

The low pass filter in my opinion was the one that I had the hardest time understanding, and probably, for many of my digital audio recorders, never bothered to used because I wanted to edit it myself and have it be recorded totally RAW and with no effects included in the audio and such, except the intensities of the audio and such (This is why when I go to concerts, I put it at the low volume so it doesn't end up distorting, compared to classroom sessions where I put it on medium, but it picks up a ton of fan noise in the background, which sucks but that's what it is sadly).

Really had to LOL about the "Batman" reference in your comment! - :D - Made my day! - Cheers!

His video clearly shows the issues is completely misunderstood. Now, beyond the arithmetics of audio, let's get.deeper into audio itself. I was educated in sample rate affects by THE Rupert Neve. If you don't know his name, you don't know audio. Look him up. Anyhow, he can show the affect of higher sample rates (frequency) based on overtones and how they can affect what we can hear directly through physics. I have heard EXACTLY what he is taking about through experience, and no, I don't claim to be a geeky audiophile. You just have to listen. Overtones above our hearing threshold affect frequencies within our hearing threshold. Those who smugly tell you "you can't possibly hear above 20 kHz" are only partly right, but significantly wrong when it comes to what those frequencies can do to other frequencies. Simpletons don't know this and say its "just audiophiles acting snobbish" when in reality they just don't have the wider experience and education on audio (analog) and physics. It's more complex than the Nyquist theory and the digital side of things. Sad that this is not more widely understood.

@@videodistro I've read things saying those are just unwanted harmonics within the amplifier or speaker, and when the file is accurately output, they are no longer present. What's your take on that?

there's a reason audiophiles dont blindly abx test these things and then use nebulous terms to describe differences between certain dacs.

@@hand__banana Bingo!

а Котельников сделал это в 1930s

I've seen a DAC that plugged into the parallel port and simply had a bunch of identical resistors soldered to the output pins. Not the most professional solution, but this was the early 1990s, and people didn't have sound cards lying about.

You think about COVOX.

That was most likely a R2R-ladder. See en.wikipedia.org/wiki/Resistor_ladder#R%E2%80%932R_resistor_ladder_network_(digital_to_analog_conversion)

Most common DACs today are not resistor ladder because they're actually quite complicated. Most are sigma-delta these days.

Yup, the raspberry doesn't have an analogue out - it uses either PWM or Sigma-Delta. However, one-bit CD-players(MASH) are quite old, and SACD is 1 bit.

inertia186 I think you'll find it's actually a time travel comedy drama with a catchy theme tune.

Oh boy.

Same nonsense as using Ultimate to mean the best when it, in fact, means final as in the Ultimate destination.

True, but if something is the best then clearly you won't need another version of whatever said thing is because it's already as good as it can be.

I'm pretty sure it's in reference to how much QFT changed how we view the universe

Same here. Just did it.

Best explained by CGP Grey: kzfaq.info/get/bejne/qKtjncKDrdHVepc.html

Also, different audience (not *that* many people are interested in this kind of stuff)

Lack of RGB.

People just want to be entertained whilst shutting their brain off. They don't get that understanding things can be entertaining too.

*MOLECULAR FOOD SCANNERS*

Particularly since PC hard drives of the period were only on the order of tens of megabytes. This was a particular issue to those trying to create CD-ROMs. Even when DVDs first came out, they held more than the common hard drive capacity, though the ratio was smaller. By about the ’00s, hard drives had overtaken optical media and left them in the dust.

@@lawrencedoliveiro9104 but there's still no cheaper way to give someone some music they might like to listen to in their car

The car I’ve been driving lately (newer than my own) has no CD player, but it does have Bluetooth and USB ports.

@@nthgth I mean, you could download it on your phone which has a cheaper storage to cost ratio

A purist will point out that it's 650 MB, not 650 Mb. Megabytes, not megabits. Huge difference. But I'm not that fussy.

Yep

I've worked with those devices before. I thought it was cool at the time to think that you could store digital audio on a video cassette and then of course ADAT came along to do multi-track.

I think people that like vinyl for everything just don't know what they're talking about, are comparing vinyl to mp3, or have terrible DACs.

I am audiophile and agree with the above, the thing is that machines are not perfect and you need great engineering to make this work well. I am a fan of CDs because they got it right from the beginning using math and the physics. LPs are cool and effortless to reproduce but have lots of flaws. A turntable is a "simple" machine, a CD player/DAC are pretty complex ones. So the difference in reproduction is in the equipment not in the theory and there is a big difference between consumer audio gear and high fi gear.

Some people can hear up to 24KHz, but only people who haven't abused their hearing AND that are under the age of 24. This extra range doesn't contribute much to the overall sound though, and most people able to hear this tiny bit of extra frequencies don't care about good music anyway.

are you saying that you only detect sound through your ears?

Some people CAN hear past 20 kHz, as it's only the human average, not a set-in-stone value. I'm one of those freaks who can hear some of it, like when some kinds of electronics are running. Nearly broke a window because of it. My ex had left some kind of air-filter/ionizer... thing in my closet when she moved out. I pulled it out to see if it worked. Plugged it in, turned it on while holding it (worst mistake ever...), and it started making one of the highest frequency noises I've ever heard. I literally threw it across the room, yanking the cord out of the socket and almost out a window. It was awful. And no, being able to hear those higher frequencies doesn't add anything to the "music experience." All the really high frequency stuff just gets drowned out 99.9% of the time. You're seriously not missing out on anything if your hearing doesn't go outside the average range, other than the occasional headache because you can hear some tiny, awful tone/sound from a coworkers' computer that literally no one else in the office can hear.

I think they are referring to the best quality records (in mint condition), played on high end equipment.

But it would still be inferior to a digital recording played back on the same equipment. Even if you had an absolutely perfect never before played record, it would only match the quality of a digital recording on the first play, the second playback would be objectively worse from the wear of the first playback, even with an optical/laser turntable in a vacuum there would be eventual wear and degradation of the record over time. Some people might prefer the sound of a record, which again is perfectly fine, I have no doubt a lot of people do and they are not wrong for preferring it, but there is no real objective benefit to a record over digital, digital is superior in just about every way.

A digital recording cannot be played back on the exact same equipment as an analog medium. In order to really compare analog a digital, what is needed is comparison of an analog and a digital recording similar in dynamic range, frequency response, etc.

I like and buy vinyl. But it's because I appreciate them as art objects and enjoy sort of the ceremony of Putting On and Listening to a Record. I, of course, also buy CD's for the same reason and the high quality sound. :P

Records tend to sound better than CDs overall because mastering was better in the days of vinyl. And they do sound the way that the artist intended because the artists approved the master. That's not necessarily true of the catalog CD that was released decades after. But there's no technical limitation keeping CDs from sounding as good, it's just that most of the time they don't. Of course not all of this is true if you only listen to newer music so your experience may be different than mine.

A discrete voltage is still analog.

@@BrianMarshall1 that's why I've had my professor at school explain to me and our class that for digital signals, digital is just a layer over analouge. We just interpret voltage levels as digital data. That's why noise/interference can be harmful to a digital signal. Even if the data is right, noise/interference can cuase the bits to flip.

nope, there is definitely spectra associated with outputting a jagged signal. ('infinite' mirrors of the baseband signal), and basically we remove all that with analog filters to get the 'perfect reconstruction'

Yes. The samples are simply snapshots of a point on a waveform in a small fraction of time. Mathematically, the only possible waveform that can pass through all those sample points is the original waveform. Thus the samples recreate the exact waveform that was sampled. Nyquist states that to reproduce the exact waveform you must have at least 2 sample points between a peak and a trough. Thus you must sample at twice the frequency you wish to be your upper limit. So CD is 44,100 samples a second to perfectly recreate frequencies below 22,050 Hz.

It took me days to get my head around that. The fact that it's recursive means there's no easy starting point - every loop needs the previous looped steps to have happened for there to be any data to make sense of/learn about.

Not really. In as much as you can't have a waveform without multiple samples, yes. But, the effect of past samples diminishes greatly in a short time due to the band limiting again. Instead of multiple vertical steps, as in the (PCM) examples in this video, DSD or PWM (same thing, different marketing) just oscillate from the extreme top and bottom values at a much much higher rate. Instead of 65536 vertical steps at 44kHz, there are only two steps but at a couple MHz.

That's one of the steps in the simpler form of PWM. Omitting lots, the error between this 0 or 1 is constantly fed back into the input to give a correct average over time (essentially making the 0s and 1s "down" or "up" commands for the drawing of the waveform). It's not as simple as just increasing the sample rate and decreasing the bit depth. I studied this stuff.

I think you're thinking of Delta-Sigma modulation, which uses successive approximation to achieve the result of a resistor ladder without the requirement of super high precision resistors. PWM doesn't have or need a feedback signal. The duty cycle determines the output. Whatever is encoded in the bitstream gets transmitted as-is and low-passed. You can actually make a DSD "DAC" with a push-pull TTL output, and an RC filter. Now if you're talking about the encoding process, then yes - you need an oscillator, a comparator (too high? too low?), and a feedback from after the filter as an input to the comp. I study this stuff too. ;-)

Yes I was - you're right.

Yeah, as I understand it, the sample rate only needs to be very slightly above double the frequency limit, but it can't be exactly double. It's an exclusive minimum.

Please remember that the sample rate on a CD is 44100 and the filter cuts all frequencies above 20Khz. The sample rate is actually well above the nyquist rate and so any sine wave that passes the filter has its phase relationship preserved.

"Audiophiles are the flat earthers of music". I love it! (I'm also stealing it...). :-)

@@zaphodsbluecar9518 You're very welcome! LOL

Not all audiophiles... Take a look at .FLAC vs mp3 so digital audiophiles would touch mp3 FLAC only 😂

@@alexschoenfelder-lopez4374 32bit/384kHz min or keep it to yourself 😒

I think most audiophiles just pretend they hear the difference between 100$ and 1000$ audio cable, even if they don’t hear or know for sure there isn’t. People really like the idea of belonging to the elite, so no other audiophile would destroy this illusion, or they all would draw themselves as pretentious liars at best or fools who spent insane amount of grands on something practically useless at worst. You need to train your ears and brain for a loooong time in order be able to analyze what you hear. This isn’t a skill you obtain passively listening music. After certain threshold even the best audio engineer in the world wouldn’t notice difference in “quality” or whatever. Not to mention that the older you get the narrower your hearing range becomes (especially it’s upper threshold)

It makes you realise that many CD's that sound bad sound bad because they are perfectly reproducing a bad recording and not because they are cd's. Then you see that the so called "hi-res" music (which isnt needed as cd audio is designed to be perfect anyway) sounds better because they used a better quality recording. So why are us consumers being messed with when buying cd's?

Yes, that's what I said in my earlier comment. Basically, if you have stair-step signal, you get about -3 dB loss at 20 kHz. With 2x oversample, the loss will be -0.75 dB, and with 4x, it's -0.18 dB. Coincidentally, an effect on the frequency response of using 50% duty cycle bars will be the same as 2x oversample, and with 25% duty cycle it's same as 4x oversample. With true oversample, though, you'll just get the aliases farther in frequency, so you can use a simpler analog filter.

Correction: with stairstep signal, the frequency response at near Nyquist frequency goes to 2/π, not to 0.

Ah, yes, I was off by a factor of two. It goes to 0 at the sampling frequency (which makes sense) and at half that you get 2/π, which is -3.9dB or so. And yeah, the point of oversampling is to get rid of the aliasing within the new wider response digitally, so then the analog filter is much easier to make.

Because you are dealing with theoretical vs. practice. It's easy to say "the low pass filter..." but low pass filters aren't perfect. Early brick wall filters rang horribly, causing artifacts to the audio as it limited it. Even today they're not perfect. Then DACs are also not perfect. It's nearly impossible to design ladder DACs these days because resistors have variances and they cannot be trimmed to an accuracy where even the two channels in a stereo DAC provide the exact same output per mono channel fed into either channel. Modern single bit DACS have their own issues, and finally, there's that filter problem again. There are also a variety of other issues, but these are some of the easier to explain ones.

There also the truth that while analog audio has much more errors and smearing than in digital realm, those "problems" can sound comfortable and even really good. That's why analog recorded albums sound (subjectively) better and why people love vinyl even today. Guitars are perfect example of this. Electric guitars get their traditionally distinctive soundscape from poorly designed amplifier circuits, sometimes enhanced by pushing them too far to create walls of analog distortion.

Because nobody makes solid-gold transistors.

@@VideoArchiveGuy Digital low-pass filters are actually perfect, within the limitations of what a real-life oscillation is. The ringing of a very steep low-pass filter is concentrated at the knee frequency, meaning it would be at 20 kHz or above for 44.1 kHz. There are three reasons why digital can be abhorred by "audiophiles". The first one is that these low-pass filters were originally done in analogue, and that caused analogue ratty distortion on the top end of a signal. Nowadays converters don't work the same way at all. They capture a very high frequency range, way outside the limit of hearing, and then apply a digital filter that doesn't cause that distortion. The second one is that a lot of people have absolutely no idea about the info in this video. I have heard the "stairstep", the "ultrasonic info", and the "minimum timing information" myths from people who really, really should have known better amongst sound engineers; the average Joe would fare way worse. The third one is that music often sounds pretty cool with the added distortions of tape, vinyl, and extra analogue stages. A lot of that distortion sounds nice! Even today, most pop music recordings have distortion added to them because, well, it sounds better that way. People would be shocked at the amount of distortion, either digital or analogue, added to almost every pop and rock album even today. However, if it sounds better to add more distortion at the playback stage, it means that maybe the producers and engineers should have added it at the production stage. But that's just a preference. The absolute proof that digital is a better medium than vinyl is that you can record a digital signal into vinyl and it will sound like vinyl. However, you can record a vinyl signal into digital, and it will sound *exactly* like the vinyl.

@@distortingjack Digitally sourced vinyl still usually sounds like digital. It is true that vinyl recorded to digital will retain character of the vinyl, with many listeners preferring the sound of a digital recording sourced from a good LP to that released as a digital recording.

Yeah by far, but 100mhz is only single channel, if you go to 2 channels it drops to 50mhz, and 3 or 4 goes to 25mhz.

soupisgdfood that is not true

Douglas Quaid I was wrong. You DO lose bandwidth due to reduced sample rate with each new opened channel, except when going from 3 to 4 channels, bit it’s due to lost resolution only. Though, that all being said, the minimum sample rate is 250Msamples/sec, which theoretically can do 100mhz but you’re really cutting it close there.

You are 100% false. The sample rate drops, not the bandwidth. 1 channel 1GS/s 2 channels 500MS/s 250MS/s on 3channels...

Douglas...Quaid...check...the...edit................................

Except that digital audio is not just sampled, it is quantised.

@@Myrtone Sampling is for frequency response, quantising does noise level. 16 bits does a very, very, very low noise level, I recall somethling like 85+db, and dithering drops that much further. Way beyond what you can discern in a typical listening environment.

Its perfectly fine to prefer the analogue stuff. While it may be objectively less identical to the original recording, that doesn't make it worse, in the same way a painting is no worse nor better than a photo.

Indeed - professionally back in the day we used U-matic to master audio albums digitally and run analogue copies from that!

Yeah, was super cool to finally learn the origin of the weird 44.1 rate

Sounds like the reverse of how the ZX Spectrum computer loaded data from a program stored on cassettes, which created sounds similar to dial-up internet that produced raster bars on the screen.

Yep. Few know that it's the reason for 44.1kHz.

Yeah that blew my mind as well ;)

I got it - Image-Line has published Xiph.org's videos to KZfaq. They must have been recorded with a screencasting tool from their KZfaq channel. But you can download the original ones for free!

interesting question. Google says 5GHz?

That is about right. The mean distance a molecule travels between collisions can be estimated as 50-100 nanometers. With the speed of sound of 340 m/s you can get your limit. If the wavelength is smaller than what a molecule runs between collisions, you can expect the regions of high and low pressure become extremely ill-defined. It gives you a limit of about 2 to 5 GHz. This is an estimate, of course. I imagine the signal becomes too noisy and attenuates too fast way before you reach this limit. (the estimate is specifically for air at room temperature and the pressure of 1 atmosphere; solids work differently and transfer sound faster)