Understanding Sample Rate

[mansr]

8 hours ago, sandyk said:

He is at least as trustworthy as you are, and far more well known than an anonymous E.E. like yourself.

I am hardly anonymous.

[mansr]

9 hours ago, Don Hills said:

 

"... the usual phase response of a shared converter player was shown. This produced a negligible phase difference at low and middle frequencies, rising to a maximum of 71 degrees at 20 kHz. As yet this has not been confirmed as being of any audible importance, unless the channels are mono-ed. ..."

- HFN&RR, March 1985.

I don't have the rest of the issue, so I could be wrong about the tester being Martin.

I measured the D-50 at 10 kHz and 20 kHz.

 

 

 

The channel offset at 10 kHz is 11.3 μs or half a sample period. At 20 kHz, it is a little lower at 10.1 μs. The schematic shows nothing external to the DAC that would explain this, so it is probably a property of the chip itself.

[mansr]

7 hours ago, Em2016 said:

Any issues with increasing clock phase noise, as you go to DSD256 and then higher to DSD512?

The crystal oscillator is ~22 MHz (512*44100), and the required bit clock is divided from that. The absolute jitter level is thus constant.

 

7 hours ago, Em2016 said:

I guess it depends on the clock used of course but do we know about clock phase noise performance of the iFi micro DACs? @jabbr

The clocks seem pretty good to me. The weak part of the iFi DACs is the analogue output section.

[adamdea]

16 hours ago, fas42 said:

 

I find it quite amusing that so many need a terribly, terribly technical reason for what is needed to get high quality sound - part of the male game to be able to hold up the Big Breakthrough over one's head in triumph, I guess.

 

What really matters is sufficient quality in the replay chain - because the inner workings of the ear/brain, and  a lifetime of learning to interpret the meaning of subtleties in the sound then does the rest. Using various multi-channel gimmicks, and fancy room setups all can help the mind grok what is intended to be heard - but it's talking the hard road; far better ROI is to fix the audible flaws; and it also means that the whole century plus of recordings history we have becomes fully accessible.

My formulation

I find it quite amusing that so many need a terribly, terribly technical reason for what is needed to get high quality sound - part of the male game which involves being over fixated with gadgets

What really matters is sufficient quality in the replay chain - because the inner workings of the ear/brain, and  a lifetime of learning to interpret the meaning of subtleties in the sound then does the rest. 

Fortunately that sufficient quality is not too high. All you need to do is to let go of the notion that ever increased kit complexity and/or expenditure is required for enjoying music. What makes the difference between one listening experience and another has far less to do with kit than you think. That's why most music lovers don;t care much about hifi. 

A whole century plus of recordings history is already accessible. It's OCD which is getting in the way.

[jabbr]

9 hours ago, Em2016 said:

 

Any issues with increasing clock phase noise, as you go to DSD256 and then higher to DSD512?

 

I guess it depends on the clock used of course but do we know about clock phase noise performance of the iFi micro DACs? @jabbr 

 

I do agree DSD512 to the iDSD sounds nice though.. but are we liking something technically 'bad'? Nothing wrong with this of course.

 

 

I haven’t personally measured the iFi— so this is general — from the crystal/physics POV, the close-in phase noise increases as the frequency increases so, all else equal, an  11Mhz clock (DSD256) will have less phase error than a 22Mhz (DSD 512) clock.

 

However the DAC probably has one clock for each frequency family ie 44.1vs48 kHz so for a DSD256 input, the BCLK is generated by divide/2 and so the phase errors should be roughly similar module the error from division (should be small)

 

I agree that the analogue output stage is probably the weakest link.

[audiventory]

9 hours ago, beerandmusic said:

I don't know how good the google translators are, but i wonder if you write in your own language and convert it using google if it will be easier to follow...probably not, but just a thought.

I use special phrase translator rather. Native and target language templates are different sometimes.

Also int he different languages some different cultural conceptions in phrase building are exists. It described in special books.

But, I suppose, practice is the best tool.

[sandyk]

6 hours ago, esldude said:

No I don't.  I remember and you can still look up stereophile and other's measures of the players.  50 khz is it.

 

 Do a proper Google search like I did originally, and you will find references to burned out tweeters etc. before they amended the filtering.

[esldude]

36 minutes ago, sandyk said:

 

 Do a proper Google search like I did originally, and you will find references to burned out tweeters etc. before they amended the filtering.

I didn't post any links earlier as I was posting from a phone. 

 

Here is the first Sony SACD player.  Down -3 db at 40 khz.  The very first SACD player from Sony or anyone.  Looks like the filtering was already amended from day one. 

https://www.stereophile.com/content/sony-scd-1-super-audio-cdcd-player-measurements-sacd-player

 

A later SACD player form Sony showing 3 db down by 50 khz.

https://www.stereophile.com/content/sony-scd-xa9000es-sacd-player-measurements

 

 

Another later Sony SACD also down 3 db at 50 khz.

https://www.stereophile.com/content/sony-scd-c333es-sacdcd-player-measurements

 

Yet another with the same result from Sony.

https://www.stereophile.com/content/sony-scd-xa777es-multichannel-sacdcd-player-measurements

 

Tweeters burned out, I don't know.  You can do that with a cassette as far as that goes.

 

100 khz response, don't see it here.  

 

dCS SACD player down 3 db at about 80 khz. 

https://www.stereophile.com/content/dcs-p8i-sacd-player-measurements

 

 

There was a Krell player that rolled off about 65 khz.  A Cary that made 40 khz (which was for the most part the norm for SACD players).  

 

Find some data, not some anecdotes.  "I burned out my tweet because the SACD goes to 100 khz and has tons of noise above." Yeah, right. 

[Spacehound]

57 minutes ago, esldude said:

I didn't post any links earlier as I was posting from a phone. 

 

Here is the first Sony SACD player.  Down -3 db at 40 khz.  The very first SACD player from Sony or anyone.  Looks like the filtering was already amended from day one. 

https://www.stereophile.com/content/sony-scd-1-super-audio-cdcd-player-measurements-sacd-player

 

A later SACD player form Sony showing 3 db down by 50 khz.

https://www.stereophile.com/content/sony-scd-xa9000es-sacd-player-measurements

 

 

Another later Sony SACD also down 3 db at 50 khz.

https://www.stereophile.com/content/sony-scd-c333es-sacdcd-player-measurements

 

Yet another with the same result from Sony.

https://www.stereophile.com/content/sony-scd-xa777es-multichannel-sacdcd-player-measurements

 

Tweeters burned out, I don't know.  You can do that with a cassette as far as that goes.

 

100 khz response, don't see it here.  

 

dCS SACD player down 3 db at about 80 khz. 

https://www.stereophile.com/content/dcs-p8i-sacd-player-measurements

 

 

There was a Krell player that rolled off about 65 khz.  A Cary that made 40 khz (which was for the most part the norm for SACD players).  

 

Find some data, not some anecdotes.  "I burned out my tweet because the SACD goes to 100 khz and has tons of noise above." Yeah, right. 

 

The Colloms and "many members" have posted anecdotes.

[Ralf11]

"many, many members"

 

The Colloms !!

[asdf1000]

6 hours ago, jabbr said:

 

I haven’t personally measured the iFi— so this is general — from the crystal/physics POV, the close-in phase noise increases as the frequency increases so, all else equal, an  11Mhz clock (DSD256) will have less phase error than a 22Mhz (DSD 512) clock.

 

However the DAC probably has one clock for each frequency family ie 44.1vs48 kHz so for a DSD256 input, the BCLK is generated by divide/2 and so the phase errors should be roughly similar module the error from division (should be small)

 

I agree that the analogue output stage is probably the weakest link.

 

Thanks jabbr and @mansr

 

The background to my question comes from Andreas Koch and Ted Smith.

 

Andreas Koch's comments here: https://positive-feedback.com/audio-discourse/raising-the-sample-rate-of-dsd-is-there-a-sweet-spot/

 

"Double DSD seemed to "fix" above mentioned compromise that was committed with single DSD by moving the noise shaper from 20kHz up to around 40kHz, well above our standard audio band of 20kHz. It also allowed for gentler and simpler output filters on the DAC. Life became a bit easier and every time that happens in audio we can expect better sonic performance. That is clearly the case with double DSD, and the price of double the data rate seems well worth it.

 

With a clear improvement from doubling the sample rate of single DSD, it seems natural and, of course, tempting to quadruple the sample rate or go even higher. We should expect the same or similar improvement as from simply doubling the sampling rate again, right?

 

Not so fast! It turns out there are physical limitations such as electronic component speeds, finite clock slopes, etc., that limit the amount of performance gain we could expect from raising the sample rate above a certain threshold. Such limitations are indeed starting to affect performance with quad DSD in D/A converters. Side effects in the form of audible noise and distortion are creeping into our audio band with quad DSD, greatly overshadowing the small benefit of the noise shaper curve starting at 80kHz. This, of course, is only apparent in D/A converters that convert the DSD signal directly into analog without any conversion to PCM or other digital filtering. To avoid this the quad DSD signal would have to be low-pass filtered and/or converted to PCM before converting it to analog."

 

Ted Smith seems to hint at similar here:

 

http://www.psaudio.com/forum/directstream-all-about-it/questions-for-ted-about-upsampling-and-fpga/page-2/#p71776

 

"Ultimately you need to actually do the conversion from digital to analog – unlike digital processing where you (more or less) get to define your own universe that final piece of hardware is restricted by physics, cost and other real world constraints.  In particular the higher frequency your clock the more noise from jitter.  As the clock frequency goes up the noise can go down because you do more noise shaping (trading noise floor in the audio band vs noise in the high frequencies).  You get 3dB more S/N over the audio band for each doubling of the sample rate.  On the other hand the noise from jitter grows as the sample rate increases as well.  The final noise floor ends up depending on the number of bits in each sample, the oversampling ratio and the jitter.  The optimum sample rate doesn’t depend much on the number of bits in each sample but for one bit audio it’s between the sample rates of double and quad rate DSD (closer to double rate DSD.)  If I go to buy (at any price) better clock crystals the jitter (in particular the low frequency phase noise) goes up with higher frequencies.  I’d like to use a 16 FS clock (45.1584MHz) but it has more noise that the 22.5792MHz clock in the DS.  Also the bandwidth of the digital switches I use is fixed and the third harmonic distortion goes up as I increase the sample rate…"

 

Is there something/s they are measuring that others haven't/can't (access to better equipment)?

[Spacehound]

1 hour ago, Ralf11 said:

 "many, many members"

 

The Colloms !!

Martin Colloms

He's a British self-proclaimed  'expert'  on  audio and the  unknowing Captain of sandyk's Pretorian Guard, (which has the regimental  motto  of  "Many  Members"). He was granted this eminent position   as he is rumoured  to have  agreed with sandyk once, but only 'statistically'.

 

Also at the top of his website he says he is an 'Expert Witness'. As he doesn't say what on we can only assume it's everything.

 

Thus he is the  perfect target for derision  

 

[Ralf11]

he isn't just self-proclaimed tho - isn't he an engineer?

 

I do understand the ultimate insult you used however "British" 

[Don Hills]

9 hours ago, mansr said:

...  At 20 kHz, it is a little lower at 10.1 μs. ...

 

Thanks, Mans. That is indeed about 72 degrees at 20 kHz. So Martin (or whoever) did measure correctly. The confusion was all mine, I mixed up sampling rate and frequency.

 

Edit:

Your D-50 still works? Last time I checked, so did mine.

[Spacehound]

14 minutes ago, Ralf11 said:

he isn't just self-proclaimed tho - isn't he an engineer?

 

I do understand the ultimate insult you used however "British" 

So he claims. AFAIK he has never been known to make anything.

 

Quite. The EU uses it too  We of course have a special name for them - "foreigners" (once "natives")  both of which are pejorative by definition.

[fas42]

7 hours ago, adamdea said:

Fortunately that sufficient quality is not too high. All you need to do is to let go of the notion that ever increased kit complexity and/or expenditure is required for enjoying music. What makes the difference between one listening experience and another has far less to do with kit than you think. That's why most music lovers don;t care much about hifi. 

A whole century plus of recordings history is already accessible. It's OCD which is getting in the way.

 

The sufficient quality is not high if you want the reproduction to be obviously 'fake'; to have a smallness about the experience, along the lines of a how a kitchen radio sounds "very small".

 

If you want the listening experience to overwhelm, to be far bigger than you are, so to speak, then the sufficient quality is very critical. It doesn't require gadgets or complexity or expenditure of money - it requires every last detail to be attended to, and sorted if necessary.

 

Think, submarine: 100% watertight, everything works beautifully; 99.9% watertight, everyone dies - that's the  type of thinking that's necessary to get audio to the right quality level.

[mansr]

31 minutes ago, Don Hills said:

Thanks, Mans. That is indeed about 72 degrees at 20 kHz. So Martin (or whoever) did measure correctly. The confusion was all mine, I mixed up sampling rate and frequency.

Well, it never hurts to double-check.

 

31 minutes ago, Don Hills said:

Edit:

Your D-50 still works? Last time I checked, so did mine.

It did today. Granted, it hasn't seen much use the last 30 years. This particular unit was manufactured in 1984. Good build quality.

[mansr]

57 minutes ago, Em2016 said:

 

Thanks jabbr and @mansr

 

The background to my question comes from Andreas Koch and Ted Smith.

 

Andreas Koch's comments here: https://positive-feedback.com/audio-discourse/raising-the-sample-rate-of-dsd-is-there-a-sweet-spot/

 

"Double DSD seemed to "fix" above mentioned compromise that was committed with single DSD by moving the noise shaper from 20kHz up to around 40kHz, well above our standard audio band of 20kHz. It also allowed for gentler and simpler output filters on the DAC. Life became a bit easier and every time that happens in audio we can expect better sonic performance. That is clearly the case with double DSD, and the price of double the data rate seems well worth it.

 

With a clear improvement from doubling the sample rate of single DSD, it seems natural and, of course, tempting to quadruple the sample rate or go even higher. We should expect the same or similar improvement as from simply doubling the sampling rate again, right?

 

Not so fast! It turns out there are physical limitations such as electronic component speeds, finite clock slopes, etc., that limit the amount of performance gain we could expect from raising the sample rate above a certain threshold. Such limitations are indeed starting to affect performance with quad DSD in D/A converters. Side effects in the form of audible noise and distortion are creeping into our audio band with quad DSD, greatly overshadowing the small benefit of the noise shaper curve starting at 80kHz. This, of course, is only apparent in D/A converters that convert the DSD signal directly into analog without any conversion to PCM or other digital filtering. To avoid this the quad DSD signal would have to be low-pass filtered and/or converted to PCM before converting it to analog."

 

Ted Smith seems to hint at similar here:

 

http://www.psaudio.com/forum/directstream-all-about-it/questions-for-ted-about-upsampling-and-fpga/page-2/#p71776

 

"Ultimately you need to actually do the conversion from digital to analog – unlike digital processing where you (more or less) get to define your own universe that final piece of hardware is restricted by physics, cost and other real world constraints.  In particular the higher frequency your clock the more noise from jitter.  As the clock frequency goes up the noise can go down because you do more noise shaping (trading noise floor in the audio band vs noise in the high frequencies).  You get 3dB more S/N over the audio band for each doubling of the sample rate.  On the other hand the noise from jitter grows as the sample rate increases as well.  The final noise floor ends up depending on the number of bits in each sample, the oversampling ratio and the jitter.  The optimum sample rate doesn’t depend much on the number of bits in each sample but for one bit audio it’s between the sample rates of double and quad rate DSD (closer to double rate DSD.)  If I go to buy (at any price) better clock crystals the jitter (in particular the low frequency phase noise) goes up with higher frequencies.  I’d like to use a 16 FS clock (45.1584MHz) but it has more noise that the 22.5792MHz clock in the DS.  Also the bandwidth of the digital switches I use is fixed and the third harmonic distortion goes up as I increase the sample rate…"

 

Is there something/s they are measuring that others haven't/can't (access to better equipment)?

What they say makes sense and agrees with my own experience. The optimal rate depends on the DAC. In the case of iFi, their weakness is a fairly high degree of intermodulation distortion in the 80-120 kHz range (roughly speaking). If the DSD noise isn't pushed clear beyond that region, the noise floor in the audible range rises. For this, DSD256 is required.

[asdf1000]

2 minutes ago, mansr said:

For this, DSD256 is required.

 

Thanks mansr but the way I read what both said, something between DSD128 and DSD256 , but closer to DSD128, is the sweetspot.

 

My bigger question then is are we enjoying more noise/distortion with DSD512 upsampling?

 

[jabbr]

1 hour ago, Em2016 said:

Is there something/s they are measuring that others haven't/can't (access to better equipment)?

 

It’s a more complicated issue than me get first appear, and individuals have biases depending on their own implementations. 

 

Yes its true that all else equal slower clocks have lower close in phase error — however what the optimal DSD rate is for SQ is a much much more complicated question.

 

There is good reason to believe that DSD512 can be better than DSD256 — so at face value don’t believe that quad rate is necessarily optimal. In my experience certainly not DSD128!

 

DSD1024 has not been widely implemented so the optimal rate could be DSD512 or perhaps DSD1024 or somewhere in between. 

 

Now that means you need logic that can itself handle high frequencies with low phase error and this logic can be either expensive or custom and so expensive

[sandyk]

Dennis

There were numerous reports of  tweeter burn out and amplifier 

 failures at the time.There was also a great deal of promotion of several types in a range of Sony 100khz tweeters at the time too that you apparently never saw either.They weren't a figment of my imagination !!!

There is helluva lot of earlier material on Google that is no longer findable, especially stuff that wasn't favourable to major  corporations.

 

 

 

 

 

 

[mansr]

Just now, Em2016 said:

Thanks mansr but the way I read what both said, something between DSD128 and DSD256 , but closer to DSD128, is the sweetspot.

That may be, but it's not an option on any DAC I know of. 

 

Just now, Em2016 said:

My bigger question then is are we enjoying more noise/distortion with DSD512 upsampling?

You'd have to measure your DAC and check how it performs.

 

I do think AK and TS somewhat overstate the problems with higher clock rates. Modern electronics handle several gigahertz with ease, and here we're only dealing with 100 MHz at most. Phase noise (jitter) is far from the only relevant parameter. If moving the noise up in frequency makes the filters perform better, perhaps overall performance is improved even if there is slightly higher jitter.

[sandyk]

1 hour ago, Spacehound said:

Martin Colloms

He's a British self-proclaimed  'expert'  on  audio and the  unknowing Captain of sandyk's Pretorian Guard, (which has the regimental  motto  of  "Many  Members"). He was granted this eminent position   as he is rumoured  to have  agreed with sandyk once, but only 'statistically'.

 

Also at the top of his website he says he is an 'Expert Witness'. As he doesn't say what on we can only assume it's everything.

 

Thus he is the  perfect target for derision  

 

An arrogant and Pompous Ass like yourself is also a perfect target for derision. As is so often the case in this forum, it's a case of " Cut down the tall poppies " Syndrome by envious little nobodies !

Perhaps you were short changed at birth too ???

[Don Hills]

5 minutes ago, sandyk said:

An arrogant and Pompous Ass like yourself is also a perfect target for derision. ...

 

I say, old chap, a little close to the knuckle, don't you think?

[sandyk]

6 minutes ago, Don Hills said:

 

I say, old chap, a little close to the knuckle, don't you think?

No I don't.

Spacecadet has insulted M.C by questioning his qualifications , when a simple Google search would have provided a great deal more information including the facts that he is a qualified E.E. , Consultant, and even a well respected speaker designer