Understanding Sample Rate

[firedog]

Beer-

Do you realize we have now changed the topic? Before we were talking about whether high res recording is more accurate than lower res recording when  both are done at a sample rate above 2X the highest frequency in the recorded material. 

We agreed the answer is no - both will result in a perfectly reconstructed waveform.

 

We are now talking about reconstruction filters, not recording sample rates. 

I said before (or was trying to say) that “no one doubts”  that different reconstruction filters can possibly produce different sounding results and that is one reason you might playback in high res. But that’s about removing artifacts easily and properly and  not about recording accuracy. 

[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.

[beerandmusic]

6 minutes ago, firedog said:

Beer-

Do you realize we have now changed the topic?

 

i never care if a topic changes...i haven't ever seen a thread on this site that doesn't (wink).  I have already conceded the nyquist theorem.

 

My agenda is the same as when i started the topic, and i have already conceded that in hind site, i would have named the topic differently....and even oversampling PCM isn't where i was going, but since we are here, I would love to see some responses....especially from mansr if he believes anything above pcm 44.1 can improve SQ....

 

[firedog]

5 minutes ago, mansr said:

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.

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.

[audiventory]

1 hour ago, beerandmusic said:

yes, and an mp3 player can sound better than the best cd player if the recording is lousy on the cd.

The point was that it is "possible" for an SACD to sound better than a CD, that the current technology itself is allows for more data, and I read SNR, bit depth, and other things as well.  That any previous obstacles have been overcome.

 

Is it your opinion that CDs "in general" sound better than SACDs "in general" with comparable hardware and recordings?

1. mp3 is lossy format. It is out of the discussion.

2. Comparison CD and SACD can not be performed technically correctly. Read details here https://samplerateconverter.com/educational/dsd-vs-pcm

 

3. Bit depth, sample rate is potential abilities only. Signal/noise ratio (SNR) is implementation matter.

In some cases, SNR may be better for 16 bit, than for 1-bit/2.8MHz.

Because there may be different quality of sigma-delta modulator.

Also need to consider dithering method for 16-bit.

And sigma-delta modulator at 2.8 MHz is complex device enough.

 

4. I stated that CD may have lesser distortions than DSD/SACD (see details in this post above).

[yamamoto2002]

"PCM signal can express continuous timing accuracy"
 

[beerandmusic]

9 minutes ago, audiventory said:

 

3. Bit depth, sample rate is potential abilities only. Signal/noise ratio (SNR) is implementation matter.

In some cases, SNR may be better for 16 bit, than for 1-bit/2.8MHz.

Because there may be different quality of sigma-delta modulator.

Also need to consider dithering method for 16-bit.

And sigma-delta modulator at 2.8 MHz is complex device enough.

 

 

So you are unable to state that your belief that SACD can be superior to CD in SQ or not?

[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.

[beerandmusic]

2 minutes ago, mansr said:

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.

 

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?)

 

Truly curious...

 

[Spacehound]

35 minutes ago, beerandmusic said:

 

i never care if a topic changes....

Oh good. Me fly fishing for trout on a Hampshire,  UK,  chalkstream, in the  very footsteps of Presidents Eisenhower and Carter,  both Bushes, plus Storming Norman of Iraq:

 

 

 

 

 

[audiventory]

6 minutes ago, beerandmusic said:

So you are unable to state that your belief that SACD can be superior to CD in SQ or not?

We can't compare formats as itself.

We can compare an audio system implementations only (recording+software+hardware).

[Fokus]

46 minutes ago, pkane2001 said:

 

Sorry, but I must disagree. For example, here's a quote from an Analog Devices tutorial on DAC process in a CD player. Highlight is mine:

 

 

The oversampling in that quote implies digital-domain filtering. The post-filtering in that quote refers to the analogue post-DAC filter.

 

In the context of audio, oversampling without digital filtering does not (significantly) change the spectrum of the sampled data, including their images. Get Octave and try it.

[pkane2001]

tbd

[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.

[Blackmorec]

It suddenly dawned on me why sampling at double the highest frequency is sufficient. May seem obvious to many but initially it wasn't to me.....All sound waves are sinusoidal. All you need is the amplitude and the number of oscillations per second (frequency) to perfectly describe the signal...whether you're recording (A to D) or replaying D to A.

 

I learned everything about ADCs and DACs from signal processing, and there you need at least 5 data points to describe a peak, a lot more would be better. But if the peaks are always sinusoidal i.e have the same, defined shape, all you need to know to describe it perfectly are amplitude and frequency.

 

Given the above, there's no reason why higher sampling rates should better describe a particular sound wave within the defined spectrum.  Ha! Got there in the end.

 

So, why is it the that higher sampling rates do sound better? A few months ago I had a demo of some YG Carmel IIs driven by AVM electronics.  I was listening to standard CDs and wasn't happy with the sound, which lacked air and acoustic resonance, replaced instead by the typical digital sting in the tail.  I complained to the dealer, who pushed a button on the amp, after which all was sweetness, light and all the air and acoustic resonance you could want, with no trace of digititis. Clearly it was unresolved components of the music that had caused the problem. As soon as they were properly resolved, away went the problem.  

So what had the dealer done? Switched to the amp's upsampling mode.

[beerandmusic]

3 minutes ago, Blackmorec said:

It suddenly dawned on me why sampling at double the highest frequency is sufficient. May seem obvious to many but initially it wasn't to me.....All sound waves are sinusoidal. All you need is the amplitude and the number of oscillations per second (frequency) to perfectly describe the signal...whether you're recording (A to D) or replaying D to A.

 

I learned everything about ADCs and DACs from signal processing, and there you need at least 5 data points to describe a peak, a lot more would be better. But if the peaks are always sinusoidal i.e have the same, defined shape, all you need to know to describe it perfectly are amplitude and frequency.

 

Given the above, there's no reason why higher sampling rates should better describe a particular sound wave within the defined spectrum.  Ha! Got there in the end.

 

So, why is it the that higher sampling rates do sound better? A few months ago I had a demo of some YG Carmel IIs driven by AVM electronics.  I was listening to standard CDs and wasn't happy with the sound, which lacked air and acoustic resonance, replaced instead by the typical digital sting in the tail.  I complained to the dealer, who pushed a button on the amp, after which all was sweetness, light and all the air and acoustic resonance you could want, with no trace of digititis. Clearly it was unresolved components of the music that had caused the problem. As soon as they were properly resolved, away went the problem.  

So what had the dealer done? Switched to the amp's upsampling mode.

 

just curious, what amp had this "upsampling mode switch" and do you know specifically what it did?

[Spacehound]

1 minute ago, Blackmorec said:

It suddenly dawned on me why sampling at double the highest frequency is sufficient. May seem obvious to many but initially it wasn't to me.....All sound waves are sinusoidal. All you need is the amplitude and the number of oscillations per second (frequency) to perfectly describe the signal...whether you're recording (A to D) or replaying D to A.

 

I learned everything about ADCs and DACs from signal processing, and there you need at least 5 data points to describe a peak, a lot more would be better. But if the peaks are always sinusoidal i.e have the same, defined shape, all you need to know to describe it perfectly are amplitude and frequency.

 

Given the above, there's no reason why higher sampling rates should better describe a particular sound wave within the defined spectrum.  Ha! Got there in the end.

 

So, why is it the that higher sampling rates do sound better? A few months ago I had a demo of some YG Carmel IIs driven by AVM electronics.  I was listening to standard CDs and wasn't happy with the sound, which lacked air and acoustic resonance, replaced instead by the typical digital sting in the tail.  I complained to the dealer, who pushed a button on the amp, after which all was sweetness, light and all the air and acoustic resonance you could want, with no trace of digititis. Clearly it was unresolved components of the music that had caused the problem. As soon as they were properly resolved, away went the problem.  

So what had the dealer done? Switched to the amp's upsampling mode.

Got nothing especially  to do with sound. It's about 'information'.

 

Nyquist/Shannon  works on bus  timetables.

[beerandmusic]

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

 

http://www.wescottdesign.com/articles/Sampling/sampling.pdf

[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.

[beerandmusic]

18 minutes ago, mansr said:

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

I don't know what a sinc filter is or how you get one of infinite length, or what any of this means (grin), but

I would equate "you can get as close as you need to" to a statement similar to "little audio difference"?

 

 

[beerandmusic]

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

[psjug]

Those of you who continue to help on this are better people than me.  Is there such a thing as infinite patience?

[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.

[beerandmusic]

1 hour ago, mansr said:

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

 

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....

 

[Spacehound]

24 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

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"