[71 dB]

Audio is not usually a simple sinusoid, at all. A simple sinusoid is 1 frequency playing only. You could probably only do that electronically. Most sounds are a range of a frequency. Things can sound better that way. That's why stringed instruments are hollow, wood, and curved the way they are, for the range and tone of a good sound.
And if you tried to connect the dots of a simple sinusoid, would you try straight lines to each dot, or do you try to make your own curve in between? Which would work better?

I think Gregorio means all music can be broken up into a collection of sinusoidals with different freiquencies, amplitudes and phase (Fourier analys).

Anyway you can connect the dots in infinite ways, but only one way doesn't contain frequencies above Nyquist frequency. Since the reconstruction filter doesn't allow frequencies above Nyquist, what comes out of a DAC is FORCED to be the correct original analog signal. The only difference is that there is also quantization noise/dither (inaudible in practise in 16 bit audio).

If you try connecting the dots with straight lines, you have "corners" at the dots when the next line segment has a different angle. those sharp corners contain infinitely high frequencies. The reconstruction filter makes this impossible.

 

[71 dB]

And “NO” more dots make absolutely no difference at all!

G

Well, they make the difference of allowing higher frequencies in the signal, but above 20 kHz frequencies the difference doesn't matter at all, because we human don't hear ultrasonic signals.

 

[gregorio]

If you show me where I said foobar was a dac, i'll be the moron instead of you.

We are talking about digitising and reconstructing a the analogue signal you bought up a player software, which doesn’t do either, hence your the moron.

Then why can't you make me understand?

Me and others have explained it to you and you’ve been provided with an easy to understand video which clearly explains and demonstrates it but you simple ignore it all and carry on repeating nonsense. So you have already answered your own question: You have NOT been “careful about which one of us looks stupid”!!

I said I'm not going to turn it on, show me where I said I did.

So that’s a “no” then, you do not know why the dither checkbox is there, oh dear.

The original file is unaltered stair steps. That's how I want it to arrive at my dac.

Make up your mind, do you want the original file unaltered OR do you want stair steps? You cannot have both because the original file is not stair steps, you would have to alter the original file to make it stair steps although you’re out of luck because there’s no way as far as I’m aware of altering the original file into stair steps before it arrives at your DAC.

Your dac does not know how to curve between samples.

That’s funny as that’s pretty much all a DAC knows. What do you think a DAC does then?

Why do you think they show you a graph with steps in the first place,

Because it’s easy to read and calculate/draw, this question is answered and demonstrated extremely simply in the video provided. Were you really not able to understand it?

if there was no difference between digitizing and recording the file identically to what it originally was?

That makes no sense! Digitising is NOT recording the file identically to what it originally was, that’s the aim of analogue recording, not digital recording. Digital audio is obviously different to analogue audio, with digital audio we do NOT “record the file identically”, we record digital data which allows the original analogue signal to be perfectly reconstructed.

Clearly you don’t even know/understand the fundamental basics of what digital audio is. That’s understandable but what isn’t understandable is that you refuse to even try to understand and just keep arguing from ignorance. How’s that “being careful about not looking stupid”?

G

 

[Audiophiliac]

This is incorrect

This is incorrect.

Ok, I suppose you could try explaining how the file stores curved analogue waves, and your dac must not have to curve them because of it.
Maybe you can tell my why they show those jagged-curved diagrams every time, if they don't exist.

 

[Audiophiliac]

The problem here is you don't understand how little you know about digital audio. You don't get it that you could learn from us.

Simply put: The DAC "knows" what shape the original analog signal had "between" the sample points because there is only ONE analog signal possible that is bandlimited to half of the sampling frequency and goes through the sample points. The reconstruction filter gives you the smooth original signal automatically, because that's how the math works. In order for this to works, we need to take samples at least twice the highest frequency in signal. That's why 44.1 kHz sampling for audio that goes up to 20 kHz.

If the dac knows what's going on between the samples, does it also already know how to play the track between 1 and 3?

 

[Audiophiliac]

We are talking about digitising and reconstructing a the analogue signal you bought up a player software, which doesn’t do either, hence your the moron.

You brought up dithering, which I don't care about, and said it's a recording technique. I told you foobar has a toggle in it for playback, and you thought that meant foobar was a dac. You are not just a tiny moron, for that.

Me and others have explained it to you and you’ve been provided with an easy to understand video which clearly explains and demonstrates it but you simple ignore it all and carry on repeating nonsense. So you have already answered your own question: You have NOT been “careful about which one of us looks stupid”!!

So that’s a “no” then, you do not know why the dither checkbox is there, oh dear.

Dither will change my output, I don't want that.

Make up your mind, do you want the original file unaltered OR do you want stair steps? You cannot have both because the original file is not stair steps, you would have to alter the original file to make it stair steps although you’re out of luck because there’s no way as far as I’m aware of altering the original file into stair steps before it arrives at your DAC.

So, the original file created by a adc chip is actually analogue?

That’s funny as that’s pretty much all a DAC knows. What do you think a DAC does then?

Because it’s easy to read and calculate/draw, this question is answered and demonstrated extremely simply in the video provided. Were you really not able to understand it?

That makes no sense! Digitising is NOT recording the file identically to what it originally was, that’s the aim of analogue recording, not digital recording. Digital audio is obviously different to analogue audio, with digital audio we do NOT “record the file identically”, we record digital data which allows the original analogue signal to be perfectly reconstructed.

So, are you now saying that the original file is stairstepped, after all? And that your dac knows what goes on in between what you tell it?

Clearly you don’t even know/understand the fundamental basics of what digital audio is. That’s understandable but what isn’t understandable is that you refuse to even try to understand and just keep arguing from ignorance. How’s that “being careful about not looking stupid”?

How about if I just went along with 'digital records exactly what happened better and with all the detail there is possible to capture, compared to analogue recordings, and it is not possible for it to be better?
This is why I brought up the guitarist playing 100k riffs per second. Would a 44.1 dac chip playing that file play all the riffs, because it knows what goes on between the samples, or would it only be playing 44.1k of the riffs? Maybe I should just say 'strums' insteaad of riffs, you can't expect people to play that fast yet.

 

[gregorio]

Audio is not usually a simple sinusoid, at all.

Audio is always a simple sinusoidal waveform.

A simple sinusoid is 1 frequency playing only.

No it’s not, a single sine wave is 1 freq playing only. I was using the term “sinusoidal” to mean a geometric shape with features similar to a sine wave (such as an analogue waveform) which is relatively simple compared to all the different geometric shapes used in the Sistine Chapel paintings.

Most sounds are a range of a frequency.

All naturally occurring sounds are a fundamental frequency (sine wave) plus various harmonics (more sine waves superimposed).

That's why stringed instruments are hollow, wood, and curved the way they are, for the range and tone of a good sound.

Some string instruments are not hollow. When they are hollow, it’s to resonate and amplify the frequencies (sine waves) produced by the string vibrating.

And if you tried to connect the dots of a simple sinusoid, would you try straight lines to each dot, or do you try to make your own curve in between? Which would work better?

Nonsense question, because the answer is neither! What actually happens is a mathematical formula is applied, for which the ONLY solution/result is the original analogue waveform.

G

 

[Audiophiliac]

Audio is always a simple sinusoidal waveform.

Then what's the difference between a tone generator playing a single frequency, and a symphony playing?

No it’s not, a single sine wave is 1 freq playing only. I was using the term “sinusoidal” to mean a geometric shape with features similar to a sine wave (such as an analogue waveform) which is relatively simple compared to all the different geometric shapes used in the Sistine Chapel paintings.

All naturally occurring sounds are a fundamental frequency (sine wave) plus various harmonics (more sine waves superimposed).

Sounds like they're getting less simple.

Some string instruments are not hollow. When they are hollow, it’s to resonate and amplify the frequencies (sine waves) produced by the string vibrating.

Nonsense question, because the answer is neither! What actually happens is a mathematical formula is applied, for which the ONLY solution/result is the original analogue waveform.

You should tell video gamers they don't need higher frame rates, because their screens are showing what was there identically to the original, so you can't beat digital video.

 

[gregorio]

Well, they make the difference of allowing higher frequencies in the signal …

Yes but I was referring to the band limited signal.

I suppose you could try explaining how the file stores curved analogue waves

We can’t explain something that doesn’t exist. The file does not store the analogue waves, that’s what makes the difference between analogue and digital audio and why digital was invented in the first place! Digital audio just stores data, which allows the analogue waveform to be reconstructed. That’s all it is, just points of data, no curves or anything else.

Maybe you can tell my why they show those jagged-curved diagrams every time, if they don't exist.

At least you’re asking questions now instead of just arguing nonsense, so that’s a step in the right direction. Unfortunately though, it’s still a nonsense question because it’s already been explained to you several times and it’s even demonstrated in the video presented!

You brought up dithering, which I don't care about, and said it's a recording technique.

It doesn’t matter whether you care about it or not, dither is already applied on every digital recording ever made.

I told you foobar has a toggle in it for playback, and you thought that meant foobar was a dac. You are not just a tiny moron, for that.

No, I never thought Foobar was a DAC. Foobar does not reconstruct/convert the data into back into an analogue waveform, which is the process being discussed, so you are “not just a tiny moron”!

Dither will change my output, I don't want that.

If you don’t want to change your output then you don’t need a DAC. The whole point of a DAC is to change the output, to convert it!

G

 

[gregorio]

Then what's the difference between a tone generator playing a single frequency, and a symphony playing?

None at all, except you’d need a lot more than just one tone generator.

Sounds like they're getting less simple.

Less simple than a single sine wave but more simple than all the different geometric shapes in the Sistine Chapel paintings.

You should tell video gamers they don't need higher frame rates, because their screens are showing what was there identically to the original, so you can't beat digital video.

Why would I tell video gamers that? I’m talking about digital audio, not digital video. Didn’t you know that?

G

 

[Audiophiliac]

Yes but I was referring to the band limited signal.

We can’t explain something that doesn’t exist. The file does not store the analogue waves, that’s what makes the difference between analogue and digital audio and why digital was invented in the first place! Digital audio just stores data, which allows the analogue waveform to be reconstructed. That’s all it is, just points of data, no curves or anything else.

Digital audio was not invented to be better than analogue, only to be able to use them for audio at all.

At least you’re asking questions now instead of just arguing nonsense, so that’s a step in the right direction. Unfortunately though, it’s still a nonsense question because it’s already been explained to you several times and it’s even demonstrated in the video presented!

It doesn’t matter whether you care about it or not, dither is already applied on every digital recording ever made.

No, I never thought Foobar was a DAC. Foobar does not reconstruct/convert the data into back into an analogue waveform, which is the process being discussed, so you are “not just a tiny moron”!

If you don’t want to change your output then you don’t need a DAC. The whole point of a DAC is to change the output, to convert it!

Still waiting for how a dac can play 100k guitar strums per second at 44.1 sample rate. If your explanation makes it that far, I'll ask how it knows how to play back each string of each strum individually, without higher than 44.1 res. I'm a big speedmetal fan, who cares about how things really sound.

 

[Audiophiliac]

None at all, except you’d need a lot more than just one tone generator.

Then it's not just a simple sinusoid, after all?

Less simple than a single sine wave but more simple than all the different geometric shapes in the Sistine Chapel paintings.

Why would I tell video gamers that? I’m talking about digital audio, not digital video. Didn’t you know that?

Oh, audio does it perfectly in the first place at 44.1.
HIgher res is for morons.
Is that why MQA is better? You don't get stuck with useless information in between the original samples?

 

[gikigill]

Lord bear me strength, the stupid and Dunning-Kruger is strong in this one.

 

[Audiophiliac]

Lord bear me strength, the stupid and Dunning-Kruger is strong in this one.

See what I have to be if I go along with you guys?
Will turning something you're looking at into connect-the-dots ever have enough dots? Some people will think with 44.1 of them, your result will be identical to the original, and if you need more, you're Dunning-Kruger. Those guys got famous for making people understand that. They should go into much greater detail.
Still waiting for how 44.1 captures and can play back a 100k strumming rate, so that we can get into 'but can you make out each string of each strum?', like you could if it were analogue.

 

[gregorio]

Digital audio was not invented to be better than analogue

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?

Still waiting for how a dac can play 100k guitar strums per second at 44.1 sample rate.

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!

I'll ask how it knows how to play back each string of each strum individually, without higher than 44.1 res.

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.

Then it's not just a simple sinusoid, after all?

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.

Oh, audio does it perfectly in the first place at 44.1.

Within the band limited constraints (0-20kHz) “yes”. As proven over 70 years ago and demonstrated in the video presented to you. How many times?

You don't get stuck with useless information in between the original samples?

I had to read this twice, it almost looks like you’ve got something right! The information between the sample points is effectively useless. It doesn’t need to be stored because it can easily be reconstructed.

G