Understanding Sample Rate

[mansr]

45 minutes ago, jabbr said:

Hmmm ... it’s often though that SDM is less intuitive than PCM

It is. SDM is based on sampling, just like PCM, and then adds heaps of additional maths on top. If you don't understand sampling according to Shannon-Nyquist, you don't have a chance of understanding SDM.

 

45 minutes ago, jabbr said:

but it’s also described as “analog”

That is fundamentally wrong.

[mansr]

17 minutes ago, crenca said:

A dirac pulse is (or does it merely approach?) a state of energy at a "point in time" (or is this incorrect?), but by definition a frequency assumes a period of time...

A Dirac pulse is a signal whose value is non-zero at one single instant and zero for infinite duration before and after. It contains all frequencies from zero to infinity throughout the infinitely long interval.

[mansr]

8 minutes ago, jabbr said:

Aside from whether a truly infinite number of frequencies can exist — as I’ve said above they physically can’t —

Whether an infinite number of frequencies can exist isn't important in practice. What matters is that they can't be infinitely high.

[mansr]

6 hours ago, beerandmusic said:

watching this my assumption posed in my last question to you was correct...it is capturing a plot of the amplitude of the composite frequency, which is in agreement with your statement that frequencies don't exist at a point of time, but in essence it is measuring an amplitude plot of the waveform.

Yes, a sample is nothing but the instantaneous value of the signal at a point in time. It tells nothing about the value at earlier or later times. A single value could be part of a signal of any frequency from zero to infinity. We can only start talking about frequency once we have multiple samples spaced over a period of time.

[mansr]

6 hours ago, beerandmusic said:

Assume 10000 different tone generators (starting and stopping at different picoseconds) all at different frequencies between 10hz and 10.1hz (e.g. similar to 10.00000001, 10.00000002, 10.0000003, but not necessarily linear in difference etc)

As has already been pointed out, starting and stopping a tone generator creates additional, higher, frequencies. Focus, to begin with, on the steady state of all the tone generators running uninterrupted.

 

6 hours ago, beerandmusic said:

Granted we would not be able to discern differences, but with only a needed 20 samples per second, would we not be able to more accurately capture with a higher sample rate?

Isn't it possible that the 20 samples per second that we capture were amplitudes that corresponded to frequencies 10.000000000021 though 10.00000000040, when in actuality during that same second the composite included those frequencies once, but all the other frequencies occurred more often, and that if we captured 40 samples per second we may have captured plots corresponding to frequencies 10.000000050 through 10.000000089 which was more accurate to reality?

The composite waveform includes all the frequencies all the time. Sampling at a rate higher than twice the highest frequency is enough to capture them all perfectly. However, with more closely spaced frequencies, we need to sample for a longer duration in order to tell them apart. A single sample tells us nothing about frequency, a few samples tells us a little, and many samples tell us a lot. Now I repeat, because this is important, those samples must be spread over a long duration. Once the sample rate exceeds twice the highest frequency component, adding more samples within a fixed duration does not improve accuracy.

[mansr]

1 hour ago, beerandmusic said:

 

What a load of twaddle. Everything that guy says is wrong, and I mean everything.

[mansr]

2 minutes ago, beerandmusic said:

that is what i meant....does mansr believe that it can improve SQ?

Why don't you ask him?

[mansr]

1 minute ago, beerandmusic said:

I am ...i knew you would be here, and i did ask you specifically about what hans said in the video about increasing sampling for filtering purposes to improve SQ..

I don't know what what he said in that video as I didn't watch more than a few seconds, which was already enough to ruin my morning. If Hans Beekhuyzen ever speaks a word of truth, it is purely by accident.

[mansr]

21 minutes ago, firedog said:

He seemed to be saying that recording in high res  made sense because it enabled the use of more successful/more easily implemented reconstruction filters in a DAC.

I switched it off after he opened by collectively insulting all engineers and saying something about "far more clever people" than Harry Nyquist. Nothing good can come after that.

[mansr]

24 minutes ago, beerandmusic said:

Forgetting Hans or your opinion of him, do you believe that either DSD or upsampled PCM can improve SQ in any way, or are you of the opinion that nothing can improve SQ above a standard CD (or non-upsampled PCM 44.1 or however you want to phrase...i think you understand my question?)

Digital oversampling is a convenient way of constructing an accurate DAC. It's an implementation detail, nothing more.

[mansr]

20 minutes ago, beerandmusic said:

quote by unknown source:  nyquist never proved that you can perfectly bandlimit a signal and in practice you can't.

A sinc filter of infinite length does exactly that. In practice, you can get as close as you need to.

[mansr]

Just now, psjug said:

Those of you who continue to help on this are better people than me.

Others reading this thread might be helped.

 

Just now, psjug said:

Is there such a thing as infinite patience?

Infinite popcorn is what this calls for.

[mansr]

11 minutes ago, Spacehound said:

THIS IS THE RELEVANT PART. Note carefully  what it says.

 

"Simply stated, the Nyquist criterion requires that the sampling frequency be at least twice the highest frequency contained in the signal, or information about the signal will be lost"

Note also that this is slightly incorrect. The sampling frequency must be greater than twice the highest frequency component in the signal. Exactly equal doesn't cut it.

[mansr]

1 minute ago, beerandmusic said:

criteria of "bandlimited signal" (like telegraph system, which the theorem was initially designed for)

So band-limit the signal before sampling. The theorem is generic. It applies to everything, including things that haven't been invented yet.

[mansr]

8 minutes ago, esldude said:

I agree he is not trying to learn. 

 

My guess from his responses is he probably isn't able to learn this.  It is beyond him.

At least not in a couple of days. To be fair, it took me a lot longer to learn it too. More like a couple of years, including all the prerequisite calculus etc.

[mansr]

6 minutes ago, crenca said:

That is my position as well.  It is not that DSD is "more accurate", it's just that the iFi Micro (or do I have the IDac2?) "likes" DSD a wee bit better than PCM - it "implements" it slightly better...

DSD128, yes. DSD64 has some issues.

[mansr]

1 minute ago, crenca said:

Makes sense.  I have not even tried upsampling to DSD64 for quite a while now - I just upsample to DSD256 since that is trivial for my system and the iFi seems to like it...

Yes, DSD256 is better still.

[mansr]

4 minutes ago, beerandmusic said:

No theorem will convince me otherwise.

I'm so sorry.

[mansr]

9 minutes ago, beerandmusic said:

EE's yawn...they are still mortal

How can you be sure? I might live forever.

[mansr]

27 minutes ago, beerandmusic said:

I believe I am still very smart.

Then start acting like it.

[mansr]

3 hours ago, firedog said:

Enjoy living with your blinders on. Living is so easy when you ignore facts and truth.

Living is easy with eyes closed
Misunderstanding all you see
It's getting hard to be someone
But it all works out
It doesn't matter much to me

-- Strawberry Fields Forever, John Lennon / Paul McCartney

[mansr]

4 hours ago, semente said:

We needa builda wall. A brick wall, to keep those alias out. Out!

And make Bob Stuart pay for it.

[mansr]

22 minutes ago, adamdea said:

Of course not, but they do have a lot to do with the question what is the minimum sample rate required to capture all the information you can hear. You need to understand the sampling theorem and know something about human hearing to answer that question. Like durrr

You can also ask the question, what is the minimum sample rate required to capture all the sound there is? The answer is roughly 500 kHz. Recordings done at 352.8 kHz are now readily available. If you examine one, you will see that they have precisely zero signal content above 100 kHz or so and very, very little above 50 kHz. This means that a sample rate of ~200 kHz is enough to capture all sound in actual music, whether we are cats or humans.

[mansr]

6 minutes ago, beerandmusic said:

I did comment on it.

i read enough of it to know that it substantiated my belief and even provided an excerpt and my reasoning.

If misunderstanding were an Olympic sport, you'd be winning the gold medal every time, summer and winter games.

[mansr]

2 minutes ago, Spacehound said:

Is there a gold  for  not getting ready on time?

They tried procrastination as an event once. The competitors never showed up.