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BlueSkyy

How many bits, how fast, just how much resolution is enough?

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Ah, yes, wrong example. :) Make that two 10KHz tones with a 1 (or 99) microsecond relative time delay. Would 48K - and (e.g.) 96K samples/second PCM recordings be indistinguishable in the analog domain?

What do you mean by "two 10 kHz tones"?

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I understand your complaint. However, item 1 is can be. It can be demonstrated. Does not mean it will be or always is. I do think he is over-hyping this problem. I have found it to be a non-issue with any music I ever cared to listen to in fact.

 

Item 2 is no one heard a difference they could demonstrate. And in fact one of the complaints about that test is the hirez samples either weren't sourced from hirez or didn't have enough high frequency content to show differences. So another group of people trying to have it both ways.

 

Thanks for the response.


Founder of Audiophile Style and Superphonica

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If anybody has blind listening test info where a lot of transients were used (say snare drums, etc.) for redbook vs. hi-zoot digital, I'd really like to see it.

 

That is the program material where I'd expect a difference (if there is a difference).

 

In the past I did my own set of percussion recordings and have been using those as test tracks. Not blind test though. But easy to compare because I can also compare to the direct sound without any recording equipment involved. Mics were same distance from the instruments as my ears are when playing.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Certainly it is not. And for 96 kHz PCM you end up doing over 10x decimation factors from the actual AD conversion stage. And again over 10x interpolation factors for the DA conversion stage. Just wasted effort.

 

 

 

Processing 384k, even tens of channels in realtime is completely non-issue these days. Really.

 

 

 

Going down >100 dB in 30 kHz to 48 kHz band is practically brickwall. It is more than 100 dB/oct, that is not "gentle".

 

Would it be possible and what would be the disadvantages of having two "knees" in the filter, say 6dB/Octave at around 20kHz and then 24dB/Octave higher up?

 

R


"Science draws the wave, poetry fills it with water" Teixeira Pascoaes

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BTW, most 96 khz and above devices don't use the opportunity to use a less steep filter. They just extend the bandwidth and keep steep filters. Lavry does use a gentler filter starting to roll off above 30 khz when doing 96 khz sampling in his gear. Maybe filtering that starts at 30 khz on DXD would be great. I take it Lavry thinks you gain nothing.

 

Easier to track phase and tighten impulse response this way with the steep filters. Shallow slopes present their own problems of course.......like huge files for one

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How many bits can dance on the head of a pin? I can't hear the difference, at +65 years of age, but we do have an entertaining thread here.

 

Keep it up! Working to better the enemy of good enough will only improve the breed. (too late for many of us) but maybe we can leave a small gift for our grandchildren.

Edited by NOMBEDES

In any dispute the intensity of feeling is inversely proportional to the value of the issues at stake ~ Sayre's Law

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Would it be possible and what would be the disadvantages of having two "knees" in the filter, say 6dB/Octave at around 20kHz and then 24dB/Octave higher up?

 

That would be just 30 dB/octave filter, with it's own kind of roll-of curve... Not so much different from other multi-pole filters...


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Ah, yes, wrong example. :) Make that two 10KHz tones with a 1 (or 99) microsecond relative time delay. Would 48K - and (e.g.) 96K samples/second PCM recordings be indistinguishable in the analog domain?

Yes.

Can you prove it?


Current system: MacMini/PC  >> Schiit Eitr (with Uptone Audio LPS-1, for 5V USB power) >>  RME ADI-2 DAC (with Zerozone 12V linear power supply) >>  Xkitz Electronics XOVER-2, 100Hz active crossover (with Zerozone 17.5V linear power supply + LDOVR LT3045-A, ultra-low noise 15V voltage regulator) >> Schiit Vidar stereo power amp to KEF LS50 speakers + Sunfire HRS12/HRS8 active subwoofer. Cables used: Canare star quad speaker cables; AQ Cinnamon USB; AQ Big Sur & Schiit Pyst RCA; Supra CAT8 ethernet cable (with JSSG) for low voltage DC power; unknown coaxial cables; standard mains power cables (with Airlink BPS1502EU, 1500VA, balanced, mains isolation transformer).

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...

as for bewildering options - I find it pretty simple: I start with redbook and avoid lossy compression (maybe the psychoacoustic models for lossy are fine, but the cost to me is about zip to have non-compressed redbook).

Now, once I have the music I want on redbook; I look at the particular things that others say sound a lot better than what I already have. I don't buy it if it costs $100 for a CDRSACDHDCDwhatever, but if it is something I really want and costs me $20 or $30 then why not?

I figure (as per some above posts I made) that I am most likely buying better recording techniques, remastering, etc. but I don't care. If it really IS bit rate or bit depth, that's fine too and eventually the world will be re-Ponocized.

 

This is basically where I am with Hi-Res. I am -very- happy with quality recordings whether they come from Tidal, a CD rip or a Hi-Res d/l. I currently run everything through Roon, up-converted to DxD (or DSD128) in HQ Player and sent to my µR-->DAC. I'm hesitant to pay more than the CD price for a Hi-res version unless that particular version is not available otherwise.

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Can you prove it?

 

I would say yes. Instead let me ask, what would convince you?

 

I can send you files as you described and let you listen to see if they sound different. They won't.

 

There are two things you have in mind which are making you think they would be different which are misconceptions about how digitally sampled audio works.

 

First that you can't discriminate in time between sample points. You are missing that if a wave starts at a slightly different time, even in between samples, it generates a different value in all subsequent samples than if it started right on a sample. The reconstructed wave is continuous and will reconstruct the waves offset by the correct amount of time even between samples.

 

The other is that higher sample rates somehow are more accurate on below 20 khz signals because they have more sample points. That isn't how it works. The higher sample rate gets you more bandwidth. Both 48 khz and 96 khz will recreate a 10 khz signal accurately enough you won't see an analog difference.

 

So what would convince you?


To paraphrase Rick James, "sighted listening is a helluva drug".

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What do you mean by "two 10 kHz tones"?

I mean two *analog* tones of 10kHz, with (very slightly) different phase relative to the recording microphone..


Current system: MacMini/PC  >> Schiit Eitr (with Uptone Audio LPS-1, for 5V USB power) >>  RME ADI-2 DAC (with Zerozone 12V linear power supply) >>  Xkitz Electronics XOVER-2, 100Hz active crossover (with Zerozone 17.5V linear power supply + LDOVR LT3045-A, ultra-low noise 15V voltage regulator) >> Schiit Vidar stereo power amp to KEF LS50 speakers + Sunfire HRS12/HRS8 active subwoofer. Cables used: Canare star quad speaker cables; AQ Cinnamon USB; AQ Big Sur & Schiit Pyst RCA; Supra CAT8 ethernet cable (with JSSG) for low voltage DC power; unknown coaxial cables; standard mains power cables (with Airlink BPS1502EU, 1500VA, balanced, mains isolation transformer).

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I would say yes. Instead let me ask, what would convince you?

 

I can send you files as you described and let you listen to see if they sound different. They won't.

 

There are two things you have in mind which are making you think they would be different which are misconceptions about how digitally sampled audio works.

 

First that you can't discriminate in time between sample points. You are missing that if a wave starts at a slightly different time, even in between samples, it generates a different value in all subsequent samples than if it started right on a sample. The reconstructed wave is continuous and will reconstruct the waves offset by the correct amount of time even between samples.

 

The other is that higher sample rates somehow are more accurate on below 20 khz signals because they have more sample points. That isn't how it works. The higher sample rate gets you more bandwidth. Both 48 khz and 96 khz will recreate a 10 khz signal accurately enough you won't see an analog difference.

 

So what would convince you?

Thanks, I was hoping for mathematical proof :) I'll consider your arguments..


Current system: MacMini/PC  >> Schiit Eitr (with Uptone Audio LPS-1, for 5V USB power) >>  RME ADI-2 DAC (with Zerozone 12V linear power supply) >>  Xkitz Electronics XOVER-2, 100Hz active crossover (with Zerozone 17.5V linear power supply + LDOVR LT3045-A, ultra-low noise 15V voltage regulator) >> Schiit Vidar stereo power amp to KEF LS50 speakers + Sunfire HRS12/HRS8 active subwoofer. Cables used: Canare star quad speaker cables; AQ Cinnamon USB; AQ Big Sur & Schiit Pyst RCA; Supra CAT8 ethernet cable (with JSSG) for low voltage DC power; unknown coaxial cables; standard mains power cables (with Airlink BPS1502EU, 1500VA, balanced, mains isolation transformer).

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In my view, the problem with using audibility as a gauge is that there are aspects of equipment performance that can be affected by out of audible range noise, such as the tweeter example that I mentioned previously and if I am not mistaken the fact that some amplifiers are negatively affected by high in level supersonic noise.

 

Besides, the cumulative result of a few tiny inaudible improvements may end up being audible.

 

R


"Science draws the wave, poetry fills it with water" Teixeira Pascoaes

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Thanks, I was hoping for mathematical proof :) I'll consider your arguments..

 

Shannon-Nyquist theorem. That is the mathematical proof.

 

View the video linked in my signature. You can skip to the 20 min 50 sec. mark, and watch about two minutes. Using analog sources and analog monitoring gear with AD/DA in between he shows you can move a band-limited square wave thru various amounts of delay between sample points and see the wave shape you get on the analog o-scope is exactly the same other than moving in time relative to a second squarewave. What more proof could you want? You have the theorem predicting something, and an analog monitoring system showing the theorem works as advertised.


To paraphrase Rick James, "sighted listening is a helluva drug".

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I mean two *analog* tones of 10kHz, with (very slightly) different phase relative to the recording microphone..

Still too vague. Do you mean two tones simultaneously (which isn't a thing) or a single tone recorded twice with a slightly different phase relative to the sampling clock (this is readily simulated)?

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This is basically where I am with Hi-Res. I am -very- happy with quality recordings whether they come from Tidal, a CD rip or a Hi-Res d/l. I currently run everything through Roon, up-converted to DxD (or DSD128) in HQ Player and sent to my µR-->DAC. I'm hesitant to pay more than the CD price for a Hi-res version unless that particular version is not available otherwise.

 

My Man!.......couldn't have said it better myself!

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In my view, the problem with using audibility as a gauge is that there are aspects of equipment performance that can be affected by out of audible range noise, such as the tweeter example that I mentioned previously and if I am not mistaken the fact that some amplifiers are negatively affected by high in level supersonic noise.

 

Besides, the cumulative result of a few tiny inaudible improvements may end up being audible.

 

R

 

This assumes the production of modal harmonics from those signals outside of our range and is really easy for you and others to test yourself.......just disconnect all the other drivers in your speakers except your tweeters with response to 50khz and play loud test tones of the frequencies in question and see if those theoretical harmonics are audible?........

 

This also assumes that the recording microphones, sources and mastering and mixdown gear were not electronically capped in range above 20khz or whatever FR you suppose proves your hypothesis. Be interesting to see what common era ADCs do in this regards?.......is that content filtered out in the analog stage, digital, both?

 

Homework assignments!.......yes!

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Still too vague. Do you mean two tones simultaneously (which isn't a thing) or a single tone recorded twice with a slightly different phase relative to the sampling clock (this is readily simulated)?

I mean two simultaneous analog tones.


Current system: MacMini/PC  >> Schiit Eitr (with Uptone Audio LPS-1, for 5V USB power) >>  RME ADI-2 DAC (with Zerozone 12V linear power supply) >>  Xkitz Electronics XOVER-2, 100Hz active crossover (with Zerozone 17.5V linear power supply + LDOVR LT3045-A, ultra-low noise 15V voltage regulator) >> Schiit Vidar stereo power amp to KEF LS50 speakers + Sunfire HRS12/HRS8 active subwoofer. Cables used: Canare star quad speaker cables; AQ Cinnamon USB; AQ Big Sur & Schiit Pyst RCA; Supra CAT8 ethernet cable (with JSSG) for low voltage DC power; unknown coaxial cables; standard mains power cables (with Airlink BPS1502EU, 1500VA, balanced, mains isolation transformer).

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Shannon-Nyquist theorem. That is the mathematical proof.

 

View the video linked in my signature. You can skip to the 20 min 50 sec. mark, and watch about two minutes. Using analog sources and analog monitoring gear with AD/DA in between he shows you can move a band-limited square wave thru various amounts of delay between sample points and see the wave shape you get on the analog o-scope is exactly the same other than moving in time relative to a second squarewave. What more proof could you want? You have the theorem predicting something, and an analog monitoring system showing the theorem works as advertised.

Thanks again. So this theorem proves that there can be no audible difference between redbook and higher resolution recordings other than different audible artifacts of DA-conversion. I suppose that's possible..


Current system: MacMini/PC  >> Schiit Eitr (with Uptone Audio LPS-1, for 5V USB power) >>  RME ADI-2 DAC (with Zerozone 12V linear power supply) >>  Xkitz Electronics XOVER-2, 100Hz active crossover (with Zerozone 17.5V linear power supply + LDOVR LT3045-A, ultra-low noise 15V voltage regulator) >> Schiit Vidar stereo power amp to KEF LS50 speakers + Sunfire HRS12/HRS8 active subwoofer. Cables used: Canare star quad speaker cables; AQ Cinnamon USB; AQ Big Sur & Schiit Pyst RCA; Supra CAT8 ethernet cable (with JSSG) for low voltage DC power; unknown coaxial cables; standard mains power cables (with Airlink BPS1502EU, 1500VA, balanced, mains isolation transformer).

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I mean two simultaneous analog tones.

 

I still think you are missing his question.

 

Which of these are you thinking about:

 

 

A: one 10 khz tone in the left channel, combined with one 10 khz tone delayed by 15 microseconds also in the left channel.

 

 

B: one 10 khz tone in the left channel and one 10 khz tone in the right channel delayed by 15 microseconds in the right channel.

 

 

A: isn't going to work like you think. Identical tones out of phase with each other super-imposed partially cancel out leaving a single reduced level 10 khz tone. This can be done, but all you see if a lower level 10 khz tone with different zero crossing points. If such tones get 180 degrees out of phase they fully cancel out. If completely in phase they add together.

 

B: can be done, has been done, they do something like it in the video I suggested. I have done it, others have done it, so what would convince you it works?


To paraphrase Rick James, "sighted listening is a helluva drug".

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Thanks again. So this theorem proves that there can be no audible difference between redbook and higher resolution recordings other than different audible artifacts of DA-conversion. I suppose that's possible..

 

it's more than just possible (given the assumptions), it's inevitable

 

but ... it IS based on some assumptions


"The overwhelming majority [of audiophiles] have very little knowledge, if any, about the most basic principles and operating characteristics of audio equipment. They often base their purchasing decisions on hearsay, and the preaching of media sages. Unfortunately, because of commercial considerations, much information is rooted in increasing revenue, not in assisting the audiophile. It seems as if the only requirements for becoming an "authority" in the world of audio is a keyboard."

-- Bruce Rozenblit of Transcendent Sound

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Thanks again. So this theorem proves that there can be no audible difference between redbook and higher resolution recordings other than different audible artifacts of DA-conversion. I suppose that's possible..

 

Of course there can be... :D

 

Theorems don't prove anything about hearing, only maths that happen in digital domain and only under certain conditions.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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There is no such thing as two simultaneous tones of the same frequency.

 

Don't tell that to the orchestra! :)


One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> router -> 2 Cisco switches connected by optical Ethernet -> microRendu -> USPCB -> ISO Regen (powered by LPS-1) -> Ghent JSSG360 USB cable -> Pro-Ject Pre Box S2 DAC -> Spectral DMC-12 & DMA-150 -> Vandersteen 3A Signature.

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