[Audiophiliac]

Sure it wasn’t, it was invented to be worse than a string and two cans. That’s why it was invented by telcos, because they wanted to cut down on all the cables and telephone exchanges and go back to the good old days. Where do you get this nonsense, do you read it somewhere or does it just come to you?

Why on earth would you still be waiting for something you’ve already been given? You were even given the mathematical proof along with the explanation!

Again, already explained to you. It “knows” because of the data in the sample points and the mathematical formula that reconstructs the original waveform from that data.

The adc chip makes a dot that corresponds to what it hears at each sample point. You could sneak stuff in between samples that it didn't find out about. Woah, someone will probably try to sneak in watching porn while recording a symphony by cutting out the sound at each sample time.

Again, it’s a simple sinusoid compared to the numerous different geometric shapes in the Sistine Chapel ceiling paintings but it is not just a single sine wave, it’s a waveform comprised of numerous single sine waves.

If that is true, does that mean digital is no good for numerous complex sinusoids in unision? What if we tried to mess up playback by using as many complex and ever-changing sinusoids as possible? Can you overload a 44.1 sample rate, so that it can't capture it all? Does the mathematical formula a dac chip comes up with change depending on what each sample is? Seriously, is it possible to fool a 44.1 rate into not actually capturing what happened between each sample?

 

[gregorio]

See what I have to be if I go along with you guys?

Nope, the “stupid and dunning-Kruger” is all your own doing. We’d much prefer if you weren’t but you seem very determined!

Will turning something you're looking at into connect-the-dots ever have enough dots?

Not in the analogue world. However, you don’t seem able to understand we’re talking about the digital world! Let me try one last time:

Let’s say we have a perfect circle that we want to store and reproduce. An analogue method would be to take points/dots on the circle and then join them up. You’re going to reproduce a jagged circle though. You could have more dots but you’d still have a jagged circle, just less jagged. The only way to get a perfect circle would be with an infinite number of dots but that’s impossible so you’d always have a somewhat jagged circle.

That is NOT what happens though, that is analogue thinking, not digital. With digital, what we do is come up with a mathematical formula that can only draw perfect circles. Then all we need to do is take two points/dots on our original perfect circle because the only perfect circle our mathematical formula can create that bisects those two points MUST BE identical to the original perfect circle. And, it doesn’t matter if you’ve got 2 dots or 2 thousand, you’ll get exactly the same perfect circle.

That’s an analogy, we’re not dealing with perfect circles, instead we’re dealing with combinations of sine waves but the principle is the same, our mathematical formula only “draws” sine waves/combinations of sine waves, so we only need 2 points (fractionally more than two points to be precise), 30 points or 3 million makes absolutely zero difference.

This is as simple as I can explain it, if you still can’t understand I’m at a loss!

G

 

[gregorio]

You could sneak stuff in between samples that it didn't find out about.

No, that is impossible, unless it has a frequency higher than 22kHz (which would be inaudible).

If that is true, does that mean digital is no good for numerous complex sinusoids in unision?

No, digital can handle any audio waveform, regardless of how many individual sine waves it contains.

What if we tried to mess up playback by using as many complex and ever-changing sinusoids as possible?

Try it, it’s easy. The most complex is white noise, all the frequencies (within the band limit) at the same time, each one varying constantly.

Can you overload a 44.1 sample rate, so that it can't capture it all?

No.

G

 

[Audiophiliac]

Nope, the “stupid and dunning-Kruger” is all your own doing. We’d much prefer if you weren’t but you seem very determined!

Not in the analogue world. However, you don’t seem able to understand we’re talking about the digital world! Let me try one last time:

Let’s say we have a perfect circle that we want to store and reproduce. An analogue method would be to take points/dots on the circle and then join them up. You’re going to reproduce a jagged circle though. You could have more dots but you’d still have a jagged circle, just less jagged. The only way to get a perfect circle would be with an infinite number of dots but that’s impossible so you’d always have a somewhat jagged circle.

No, analogue recording does not make dots, it actually records the sound with time. You can say the recording is smeared, because recorders and your medium are not perfect, but digital still has that problem too. Congratulations on realizing that without infinite points, you get jagged. What's better, though, 44.1k dots, or 96k dots?

That is NOT what happens though, that is analogue thinking, not digital. With digital, what we do is come up with a mathematical formula that can only draw perfect circles. Then all we need to do is take two points/dots on our original perfect circle because the only perfect circle our mathematical formula can create that bisects those two points MUST BE identical to the original perfect circle. And, it doesn’t matter if you’ve got 2 dots or 2 thousand, you’ll get exactly the same perfect circle.

I like trying to fool digital playback. I'm going to record diagonal lines so that they all sound like circles on your gear, and everyone thinks it sounds like how Dunning-Kruger thought people should be studying.

That’s an analogue, we’re not dealing with perfect circles, instead we’re dealing with combinations of sine waves but the principle is the same, our mathematical formula only “draws” sine waves/combinations of sine waves, so we only need 2 points (fractionally more than two points to be precise), 30 points or 3 million makes absolutely zero difference.

So, it sounds like what you're getting at, is that analogue used too many samples, instead of using a chip to do math to play it perfectly. Which chip does the math, the adc or your dac? Sounds like the dac chip. Why don't they put the formula in the adc, in case someone tries to play something that needs a different formula? Then you could just steam the bits to the dac, without having to worry about it knowing the right formula's. Oh no, adc chips only listen to what's playing at the sample rate points you choose, they just sit there in between. Ha, I'll make a recording that teaches them a lesson.

 

[gregorio]

Congratulations on realizing that without infinite points, you get jagged.

Though not in digital audio.

What's better, though, 44.1k dots, or 96k dots?

In the digital circle analogy, they’re both exactly the same as 2 dots. If you’re talking about digital audio and samples, then they’re both the same but in practice in some situations 44.1kHz is better.

Shame you don’t seem to have understood even the simplest of explanations and just presented another batch of nonsense!

G

 

[Audiophiliac]

No, that is impossible, unless it has a frequency higher than 22kHz (which would be inaudible).

What if a note starts before the sample time comes? Your recording chip will be late capturing it, and will miss part of the plastic pick first tapping the string. It won't all be captured and can't be replaced by math.
Yeah, let's stick to what's better for adc chips, then, if you think your dac chip, (which all produce different results), plays back everything between samples without being told what was there, perfectly.
Since all dac chips sound, and even measure, differently, how do we know that yours is the one that's perfect, and all the other ones are playing it wrong?
Same goes for adc chips. If only we could find a way of recording continuously, we wouldn't have to worry about samples, to make things more convenient. People are into turntables because they think digital playback is all Dunning-Kruger.
How can you make a adc chip that makes 44.1khz dots per second sound even better, since we'd need an infinite number?
Then we can ask gregorio which dac chip he uses, since he got the one that really sounds perfect. Then we just have to find out how to get the data to it perfectly, and build the best analogue stage ever built for it. We'll be so stupid if it's not his gear at 44.1 already.

 

[VNandor]

@Audiophiliac
I have some questions for you. To start with, do you think a DAC outputs "stairsteps"? If so, is there anything that would make you reconsider this belief of yours?
Do you think an ADC can only capture transients if it samples it right as the transient peaks? Is there literally anything that could convince you otherwise?

 

[gregorio]

What if a note starts before the sample time comes?

Same as what happens if a note starts before you press record on a tape machine. If you mean if a note starts between sample points, then it’s captured and reproduced perfectly, as demonstrated and proven in the article already presented to you!

The rest if your post is just more of the same dunning-Kruger nonsense!

G

 

[Audiophiliac]

@Audiophiliac
I have some questions for you. To start with, do you think a DAC outputs "stairsteps"? If so, is there anything that would make you reconsider this belief of yours?
Do you think an ADC can only capture transients if it samples it right as the transient peaks? Is there literally anything that could convince you otherwise?

I expect a dac to output stair steps of recreations of the samples captured by the adc. The prospect of money for it could make me try telling other people it will sound perfect at 44.1khz. No, I'm not a guy like that, so nope.
What does an adc do to capture what's between it's samples?
dac chips only ever play what wasn't actually going on between samples.
That's why people with turntables think digital audio people will only ever be the people Dunning-Kruger were fans of.

 

[VNandor]

What does an adc do to capture what's between it's samples?

Do you actually want an answer to that or you have already made up your mind on how digitizing an analog signal works? What could convince you that an ADC captures information of what's happening between the samples?

 

[castleofargh]

@Audiophiliac
Almost everything you've been saying about digital audio is false, yet you talk very confidently about it. So of course we all think about Dunning-Kruger and the overconfidence shown by people too ignorant about a subject to even realize how ignorant they are.
If it's any comfort, you're not the first, nor the last one to invent a false intuitive model for a non-intuitive system. The mistake is completely understandable, but is still very much a mistake. If you don't like the Dunning-Kruger reference, a good start would be to second guess your own beliefs once several people have tried to point out how wrong they are.

Here is a short introduction to digital from an actual professor. Maybe you'll be tempted to trust that more than us? Your own ideas so far suggest that digital conversion is not converting anything. Which should be a clue of how misinformed you are because both ADC and DAC have the C of converter in them.


@gregorio, you're wasting your time. He has already declared you the enemy and won't trust anything you say. I'm not sure that he'll be willing to learn from anybody else, but I'm sure he won't learn or believe anything from you.

 

[Audiophiliac]

Do you actually want an answer to that or you have already made up your mind on how digitizing an analog signal works? What could convince you that an ADC captures information of what's happening between the samples?

Not only sampling.

 

[Audiophiliac]

@Audiophiliac
Almost everything you've been saying about digital audio is false, yet you talk very confidently about it. So of course we all think about Dunning-Kruger and the overconfidence shown by people too ignorant about a subject to even realize how ignorant they are.
If it's any comfort, you're not the first, nor the last one to invent a false intuitive model for a non-intuitive system. The mistake is completely understandable, but is still very much a mistake. If you don't like the Dunning-Kruger reference, a good start would be to second guess your own beliefs once several people have tried to point out how wrong they are.

Does that mean digital audio doesn't use samples, and just plays everything perfectly? That's how I would be wrong.

Here is a short introduction to digital from an actual professor. Maybe you'll be tempted to trust that more than us? Your own ideas so far suggest that digital conversion is not converting anything. Which should be a clue of how misinformed you are because both ADC and DAC have the C of converter in them.

I don't want to watch a video right now, but the title says "Introduces the fundamental elements of Sampling and Reconstruction from a signals perspective. Explains the Nyquist sampling rate and the effect of aliasing." If reconstruction is needed, it is not the actual. Anti-aliasing is a video game technique that helps diagonal lines from looking so much like Dunning-Kruger. They have the benefit of theoretically knowing what things look like to help with that.

@gregorio, you're wasting your time. He has already declared you the enemy and won't trust anything you say. I'm not sure that he'll be willing to learn from anybody else, but I'm sure he won't learn or believe anything from you.

I'm not the one saying digital playback is already perfect on everybody's systems, I'm the one who is the enemy who doesn't believe.
I didn't think people joined audio forums if they thought their gear was playing perfectly in the first place.
I guess if you were just a kid who was a bigshot, you could try.

 

[71 dB]

What does an adc do to capture what's between it's samples?

That's a good question for someone getting into the science of digital audio. The simple answer to this is that an ADC doesn't need to "do" anything between the sample points, because the signal to be sampled is properly bandlimited. This means it is mathematically "known" what the signal does between the samples. It can't do anything surprising, because the limited bandwidth doesn't allow that. This is the essence of the sampling theorem behind digital audio.

Mathematically this is a bit like having a parabola completely defined everywhere by knowing only three points of it. Music is massively more complex "curve" than a parabola and that's why we need 44100 sample points for every second of music, but it is enough the same way knowing just three points of a parabola is enough. Try telling mathematicians you actually need 10 or 100 points of a parabola to define it "more accurately" and they will tell you to study math just as we are telling you to educate yourself about digital audio.

 

[VNandor]

Not only sampling.

I'm not sure what you mean by that. An ADC indeed does more than just sampling in order the capture information between the samples as accurately as possible. For example, a good ADC will use filtering and dithering to help with properly capturing an analog signal. I'm asking again. Is there anything that could convince you an ADC captures information between its sample points or not?