Understanding Sample Rate

[jabbr]

15 minutes ago, beerandmusic said:

The more i read, the more i believe that i would want to sample more than just twice the highest frequency.....unless you just want "good enough".

 

That’s fine. A certain degree of over engineering is reasonable. So let’s say record at 24/96 or 24/192 ... but recording beyond that is really really hard pressed to justify any benefit (unless you were doing a ton of post-processing — even then)

 

On the other hand, upsampling for the purpose of improving the DAC is totally reasonable and an excellent technique — in fact the argument that you don’t need it would be the harder one to justify.

[beerandmusic]

9 minutes ago, jabbr said:

 

That’s fine. A certain degree of over engineering is reasonable. So let’s say record at 24/96 or 24/192 ... but recording beyond that is really really hard pressed to justify any benefit (unless you were doing a ton of post-processing — even then)

 

On the other hand, upsampling for the purpose of improving the DAC is totally reasonable and an excellent technique — in fact the argument that you don’t need it would be the harder one to justify.

 

I know many people here upsample everything to quad dsd and suggest it is notably better.

 

I really don't understand what band limited is, but even when i gave the example of the 10000 generators, MANSR stated a needed criteria of upsampling higher that i didn't really understand.

 

Regardless, I think i will upsample all my pcm (which i rarely listen to anyway) to 192K...its free to do anyway, so why not...

 

My main objective for this thread really is a totally different issue anyway, and really had to do with why an SACD sounds better than a CD, and i thought everyone believed that, but apparently not.......i guess i need to go back to living in my thread "is everything debatable". 

 

 

 

 

[davide256]

7 minutes ago, beerandmusic said:

another white paper on nyquist and the theorem was for telegraph systems and should not be taken at face value...as it is for bandlimited signals.

 

The more i read, the more i believe that i would want to sample more than just twice the highest frequency.....unless you just want "good enough".

 

http://www.analog.com/media/en/training-seminars/tutorials/MT-002.pdf

I'm not convinced at this point that increasing sample rate does anything other than make DAC  manufacture easier for good sounding results with weaker source gear. Bit density matters because it determines dynamic range.  Every source solution improvement I make, CD's sound better.  The exception is that 16 bits is inadequate for lower volume instruments  on large orchestra.

 

5 minutes ago, beerandmusic said:

 

I would hate to think things are implemented for any reason other than SQ?

So based on that statement, i will assume that oversampling can improve SQ....

 

 we all have differences in ear training and preference. I find oversampling to be the equivalent of airbrushing a photo; makes it prettier at the expense of truth in detail.

[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.

[beerandmusic]

1 minute ago, mansr said:

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.

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

[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.

[beerandmusic]

3 minutes ago, mansr said:

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

 

so you can band-limit (whatever that means -grin), or you can just increase the sample rate?

 

I agree with JABBR, WHY NOT upsample?

 

[Fokus]

52 minutes ago, beerandmusic said:

another white paper on nyquist and the theorem was for telegraph systems and should not be taken at face value...

 

The theorem predates Shannon and Nyquist with many decades, putting it firmly before the era of electrical communications. It belongs to the area of pure mathematics and was not designed or invented. It was discovered, then proven.

 

[beerandmusic]

3 hours ago, yamamoto2002 said:

 

If bit depth is infinite, it is possible.

 

I calculate real world example about 44.1kHz 16bit PCM data of 1/100th of second (441 samples).

It can store the difference of

1000.0000000 Hz from 1000.0000010 Hz but

1000.0000001 Hz signal is rounded to 1000.0000000 Hz.

By increasing bit depth to 24bit, frequency precision increases by 256 times.
 

 

why would you need 44.1khz sample if only frequency is 1000hz?  wouldn't you just need a sample rate of 2.1K?

and why would you need an infinite bit depth?

[beerandmusic]

17 minutes ago, Fokus said:

 

The theorem predates Shannon and Nyquist with many decades, putting it firmly before the era of electrical communications. It belongs to the area of pure mathematics and was not designed or invented. It was discovered, then proven.

 

proven for band limited signal

[esldude]

9 minutes ago, beerandmusic said:

proven for band limited signal

Do u understand what band limiting is?

[adamdea]

1 hour ago, jabbr said:

 

That’s fine. A certain degree of over engineering is reasonable. So let’s say record at 24/96 or 24/192 ... but recording beyond that is really really hard pressed to justify any benefit (unless you were doing a ton of post-processing — even then)

 

On the other hand, upsampling for the purpose of improving the DAC is totally reasonable and an excellent technique — in fact the argument that you don’t need it would be the harder one to justify.

Hmm alot of people go along with recording up to 96khz (but probably still distribution at 44.1) but quite a few people argue that 192 is actually too much. The problem is that you are actually letting in a whole lot of spuriae which might be better filtered out and in any event not all the editing and production tools work at that rate anyway. 

There was a case for the possibility that a bit more wriggle room over 44.1 was needed. No one was really arguing for 192 because whilst its possible to speculate that 44.1fs filters might have audible problems, the same is difficult even for a fertile imagination with fs 96 khz filters because there is so little information at the transition band and hearing 40 odd khz tones beggars belief.

The problem is that people who tended to claim that 96khz fs actually sounded better then had a habit of thinking that 192khz fs sounded better (no surprise) and so on 

As for upsampling well you need that for a digital filter whether you call it a feature (as some dacs do) or just get on with it (as most of them do). 

[firedog]

1 hour ago, beerandmusic said:

 

I would hate to think things are implemented for any reason other than SQ?

So based on that statement, i will assume that oversampling can improve SQ....

 

constructing an accurate DAC is about SQ. It can be less expensive to build an oversampling DAC. 

[adamdea]

30 minutes ago, beerandmusic said:

proven for band limited signal

So unfortunately that must mean that dsd doesn't work either.

[firedog]

15 minutes ago, esldude said:

Do u understand what band limiting is?

He apparently has an obsession with his idea of resolution and accuracy and has almost dropped it, and then goes looking on the net for material he doesn't understand so that he can come back and say what we've been telling him is wrong. 

 

It's pretty much time to give up responding to him. 

 

And yes, beerandmusic, there are conditions for the Shannon -Nyquist theorem that are never actually met in real life. That means nothing as we can get so close to the conditions that the gap is meaningless.

If you don't believe that, I suggest you stop using every bit of technology you own, and never get in a car or on a plane or train. It's all based on "almost" getting there in calculations.

NASA got to the moon using calculations made by a slide rule, because it was "accurate enough" that in real/practical terms you could call it 100% accurate.

[firedog]

4 minutes ago, firedog said:

 

mistake

[adamdea]

2 minutes ago, firedog said:

He apparently has an obsession with his idea of resolution and accuracy and has almost dropped it, and then goes looking on the net for material he doesn't understand so that he can come back and say what we've been telling him is wrong. 

 

It's pretty much time to give up responding to him. 

 

And yes, beerandmusic, there are conditions for the Shannon -Nyquist theorem that are never actually met in real life. That means nothing as we can get so close to the conditions that the gap is meaningless.

If you don't believe that, I suggest you stop using every bit of technology you own, and never get in a car or on a plane or train. It's all based on "almost" getting there in calculations.

NASA got to the moon using calculations made by a slide rule, because it was "accurate enough" that in real/practical terms you could call it 100% accurate.

It seems to me that one way of looking at the impossibility of perfectly bandlimiting is to compare with pi- you can;t actually write down all the digits but you can compute them to whatever degree of precision you require in an application. 

Rob Watts is on a mission to keep bringing out a new dac or two every year with more filter taps. It could go on forever

[esldude]

15 minutes ago, firedog said:

He apparently has an obsession with his idea of resolution and accuracy and has almost dropped it, and then goes looking on the net for material he doesn't understand so that he can come back and say what we've been telling him is wrong. 

 

It's pretty much time to give up responding to him. 

 

And yes, beerandmusic, there are conditions for the Shannon -Nyquist theorem that are never actually met in real life. That means nothing as we can get so close to the conditions that the gap is meaningless.

If you don't believe that, I suggest you stop using every bit of technology you own, and never get in a car or on a plane or train. It's all based on "almost" getting there in calculations.

NASA got to the moon using calculations made by a slide rule, because it was "accurate enough" that in real/practical terms you could call it 100% accurate.

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. 

 

[firedog]

21 minutes ago, adamdea said:

 

Rob Watts is on a mission to keep bringing out a new dac or two every year with more filter taps. It could go on forever

But at some point he may decide that he’s gotten to the point that adding more taps doesn’t actually improve the sound in practice. 

[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.

[adamdea]

3 minutes ago, firedog said:

But at some point he may decide that he’s gotten to the point that adding more taps doesn’t actually improve the sound in practice. 

errm maybe. I think that if we were going to get there we would have got there. 

[audiventory]

24 minutes ago, firedog said:

But at some point he may decide that he’s gotten to the point that adding more taps doesn’t actually improve the sound in practice. 

 

More taps need to deeper alias supression and steeper transient band (between pass and stop bands of the filter).

[jabbr]

3 hours ago, beerandmusic said:

I know many people here upsample everything to quad dsd and suggest it is notably better.

 

I upsample to DSD512 to feed to the iFI Micro DAC. This works great unless ... you start with DSD256 recordings in which case it chokes my system.

 

Having higher and higher res recordings just for the sake of it isn't necessarily better, it becomes worse.

 

But year upconverting to DSD works absolutely great for DACs that prefer DSD (lots and lots). In the same way upsampling PCM works great for DACs that prefer PCM (less but for example Phasure NOS1a)

 

So yeah I use @Miskas HQPlayer and @PeterSt XXHE to upsample/upconvert on the fly and they are absolutely great and I use @audiventory AuI for offline conversion (DSD ISOs -> DSF in particular!) --- but not for the reasons you are concerned with -- they are great because they make the DAC work better. This has to do with the DAC output filter. Upsampling does not add information to the system but it makes the sound coming out of the DAC better.

[Spacehound]

1 hour ago, firedog said:

But at some point he may decide that he’s gotten to the point that adding more taps doesn’t actually improve the sound in practice. 

It's not the number of taps the guy who pays him is counting.

[jabbr]

2 hours ago, adamdea said:

Hmm alot of people go along with recording up to 96khz (but probably still distribution at 44.1) but quite a few people argue that 192 is actually too much. The problem is that you are actually letting in a whole lot of spuriae which might be better filtered out and in any event not all the editing and production tools work at that rate anyway. 

 

Yeah if the master is at 24/192 then that's what i'd like and if at 24/96 then that, or if the master is at DSD64,128 or even 256 then that's what I'd like. I have no control of how the master is created but I'd like the recording as close to the master as possible/if possible and I can handle any conversion I want to do from there.

 

You know @beerandmusic, just consider obtaining source material (music) as above, and use software as above to convert to whatever format your DAC prefers ... and don't worry -- its all good