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Launch of Pono Studio Headphones
Old 7th May 2014
  #4741
Here for the gear
 

Quote:
Originally Posted by bandpass View Post
No, as JJ said, think about how they're reconstructed: in all cases, a quantised voltage (be it to 2, 5, 65536, or any number of levels) is LPF'd to give the analog output. The principles are exactly the same.
why do you talk about 'quantised' ?

there's no quantisation/discretisation: it's more or less (1 or 0)
like an analog signal up and down if you prefer...

musically
Old 7th May 2014
  #4742
j_j
Lives for gear
Quote:
Originally Posted by AudioTouch View Post

yes it's a language difficulty, abuse even:

...


but physically it's not hard to understand that the principles are quite different:
one is representation of amplitude at a given time (discretisation)
the other just indicate if it's more or less (1 or 0) : whence 1-bit coding (DSD/DST/PDM...?)
Is there some reason you are being both professionally insulting, again hiding behind an anonymous mic, and also spreading utterly bizzare, nonsensical quackery about PCM?

Yes, one is the amplitude, represented in many levels, at a given point, the other is the amplitude measured in ONE OF TWO LEVELS.

Same process, different time resolution, different amplitude resolution.

PCM is pcm. No difference. Both are quantized. To different amounts. That's all.

Oh, and by the way, the PULSE (signal) is what is being CODED by the signal that goes down the transmission line. I don't care what the Wiki says, if that's what it actually says, it's not even historically accurate. Sorry, worked where they did that original work, read the original BSTJ papers. Obviously not WHEN they did that work, of course. (Just to stifle more defamation based on supposed age there.)

The pulse is what is measured (in terms of amplitude), be it 65536 levels or just 2. The code is the digital representation of how it is sent down the transmission line. If you're reading something else, it's wrong. That's what happens when you read "crowdsourced information". Go to Rabiner and Gold, "Digital Signal Processing", or Jayant and Noll "Waveform Coding" if you need a definitive reference. While you're at it, bear in mind I know all four of those authors and worked with 3 of them.

You would seem to be determined to spread some ridiculous nonsense about PCM here, and you're demonstrating just why audio technology is almost always 30 years behind what the art knows.
Old 7th May 2014
  #4743
j_j
Lives for gear
Quote:
Originally Posted by AudioTouch View Post
why do you talk about 'quantised' ?
Because that's exactly what you have, a 1 bit quantiser. Or in American English, quantizer.

No more, no less. There is no "more natural" or "less natural" here, such attempts at describing the system are hopeless oversimplification at best, and are more likely to mislead (as it appears you have been mislead yourself in some fashion).
Old 7th May 2014
  #4744
j_j
Lives for gear
Quote:
Originally Posted by TonyVegas View Post
Are you guys seriously still talking about this?
Is there a point to your interjection, or do you just want to have an argument?

(Sorry, Monty Python seems appropriate here.)
Old 7th May 2014
  #4745
Airwindows
 
chrisj's Avatar
Quote:
Originally Posted by AudioTouch View Post
why do you talk about 'quantised' ?

there's no quantisation/discretisation: it's more or less (1 or 0)
like an analog signal up and down if you prefer...

musically
That is quantized. It's 'on' or 'off', 'more' or 'less', 'up' or 'down'… same principle by which dithered lower-frequency PCM works. I'll specify 'dithered' because it is possible to do 'PCM digital audio' without dithering, but it's not as good and you can make an argument that it's not correct without it—that undithered PCM is NOT even real digital audio, not properly anyway.

You could think of 1-bit SACD as super-dithered? If your idea is that with SACD thanks to the mechanism you can go to 'continuing output voltages' between the 'stairsteps' of PCM, it's pretty important to remember that SACD can only do that because it works like a (not all that optimal) form of dither. Yeah, it's bouncing around the desired value like mad and giving an output value that is 'like' an analog output value if you even it out, but PCM at lower frequencies does that too.

You can get a perfectly continuous flat output out of PCM 'between' two 'stairstep' values so long as you have any sort of reconstruction filter (either normal, or the Pono type: the Pono type is more effective at suppressing artifacts because it also suppresses valid high frequency information too). All you gotta do is rapidly oscillate between the two stairstep values at the sampling frequency, and that's telling the output filter to produce a steady flat line 'between' the possible values. You can get any position between the 'bits' by varying the duty cycle of each 'bit', just like in SACD.

The reason this doesn't work in practice to make every PCM audio encoding sound the same as SACD is, the sample rate gets reduced to make up the word length. As the sample rate goes down it becomes easier to hear the artifacts you get, whether they're 'birdie' noises like with SACD or a plain wash of noise like with TPDF dithered PCM audio. It's always about how much information you're throwing at the task, whether it's through SACD or PCM. I'm not sure if anyone's made examples of what low-bit DSD would sound like? I think it's safe to say the answer is 'bad', just like low-bit PCM. These things converge to the same point, using pretty much the same mechanism, just weighted a little differently.

I might be making a hash of the explanation, for which I apologize. I'm trying to reach guys who are skeptical of the basic concept here. JJ isn't getting anywhere just insisting DSD is PCM. I hope I've filled in some of the background as to WHY and how DSD/SACD is still PCM.

Think of it as a spectrum from,
-truncated PCM at fixed word length (any word length you like)
-dithered PCM making the word length more ambiguous w.r.t what signal values can be represented
-one-bit PCM where the signal values are so ambiguous that there's no reasonable way to process it without consolidating the information somewhat.

Every increase in ambiguity to what the signal value might be, increases the granularity of what word length you'd NEED to use to capture what's happening. And lastly:

-if you use reconstruction filtering on the DAC you get to use the sample values to reconstruct signal levels 'between the stairsteps' of what's recorded in the samples, just like DSD reconstructs signal levels between '1' and '-1'. It's the same, exactly the same. It's just a matter of how you go about it.

And THIS is why I think it's fascinating Pono is (purportedly) using a moving average filter that will literally smooth out this granularity in the crudest, most direct way possible, localised to each instant of sound. What they're doing will get the absolute maximum out of this process at the cost of filter frequency response.
Old 7th May 2014
  #4746
j_j
Lives for gear
Chrisj, that's a good start, but noise shaping is different than dither. Both should be present in an oversampled system, of course, but the way that noise is moved around in a noise-shaped system is via a feedback system that forces the error to take on a specific spectrum. The fact there is feedback in the encoder can confuse people quite a bit in their expectations of what comes out of the decoder, too.

You certainly have quite a bit right there, and you seem to be converging on the basics.

JC Candy, OJ Benjamin, Structure of quantization noise from signma-delta modulation. IEEE Transactions on Communications 29(9), 1316–1323 (1981).

RM Gray, Spectral analysis of quantization noise in a single-loop sigma-delta modulator with DC input. IEEE Transactions on Communications 37(6), 588–599 (1989).

Those might help, if you can get access to them. Jim Candy (JC Candy) and Joe Condon (who is not referenced here) came up with Sigma-Delta, later known as Delta-Sigma, and then as DSD and a variety of other things. Steve Noorsworthy also has some publications out there, but I don't have a handy reference.

An analysis of it can be found in Jayant and Noll's "Waveform Coding" under the heading "A*PCM".

Interestingly, modern cell phones use noise-shaping as well as DPCM and a variety of other techniques to get absurdly low rate, absurdly low quality audio. Noise shaping helps at any level.
Old 7th May 2014
  #4747
Gear Guru
 
Sounds Great's Avatar
 

Quote:
Originally Posted by Sounds Great View Post
These are nothing more than a continuation of the same discussions we are having here, with the same bias.
So, the big question is..
Quote:
Originally Posted by walter88 View Post
"Why doesn't Pono get behind DSD?" Am I close?

No, the question is does it sound better.

But your question is one that I posted when I first joined the conversation.

I can only guess. But Beta was a far better format than VHS, yet when you got Sony backing you, all bets are off.
Old 7th May 2014
  #4748
Airwindows
 
chrisj's Avatar
Quote:
Originally Posted by j_j View Post
Chrisj, that's a good start, but noise shaping is different than dither. Both should be present in an oversampled system, of course, but the way that noise is moved around in a noise-shaped system is via a feedback system that forces the error to take on a specific spectrum. The fact there is feedback in the encoder can confuse people quite a bit in their expectations of what comes out of the decoder, too.
Yes, absolutely—if I'm not mistaken, it's the feedback that constitutes noise shaping? I know I can make things that act like dithers with or without properly random noise (with different balances of artifacts and noise colors, much like UV22) or things that act like noise shapers (my floating point 'dither' which is only a noise shaper as you can't set a noise level against a floating point wordlength reduction with a varying mantissa).

The trouble is, this stuff so rapidly gets impenetrable to casual bystanders. Is the idea of a 'spectrum from limited rigid specifying of a signal output voltage, to ambiguous representation by averaging out a lot of more fleeting impressions' useful? I'm just trying to explain why DSD is part of the PCM digital audio continuum. It's easiest to call what it does, 'dither', in that it uses statistical density of many brief impressions to generate what seems like a stable output, just as PCM dither suppresses unwanted digital artifacts.

'Cos the thing is you can make the one act like the other. Back in the day I did a noise shaper, 'Ten Nines', that gave something like -160 dB noise floor at 20 hz (while generating an outlandish amount of noise around 20K) from 16 bit 44.1K PCM. All these things can be manipulated if you try. Ten Nines 'dither' would probably sound amazing on Ponos…
Old 7th May 2014
  #4749
Here for the gear
 

Quote:
Originally Posted by j_j View Post
Is there some reason you are being both professionally insulting, again hiding behind an anonymous mic, and also spreading utterly bizzare, nonsensical quackery about PCM?
no personal attack here: just debate for understand ; I did not make high studies... I just went to the school of life...
(1/0 - up/down) of course it is schematic/simplistic but it's not so far away from the concept!

citation:
"The 1-bit A/D converter has become the mainstream recording technology, even in conventional PCM recording. However, in the PCM technique, digital low-pass filter calculations and decimation processing are used to downconvert to the required sampling frequency. This processing discards the high-frequency components present in the original signal. Furthermore, PCM playback uses interpolation processing and noise shaping. These can subtly modify the sound quality since they are implemented as digital processing." (See figure at https://images.indiegogo.com/file_at...png?1397815086)
is it true or not?

other citation:
"Infinite resolution of the bitstream: In analogue modulators the bitstream is sampled, but not quantized. It still has an infinite resolution just like the analogue signal has. It is superimposed by noise of course, but the more noise you filter out (the lower the low pass filter's frequency limit is), the more of the original resolution is usable - (theoretically) without limitation. "


Finally:
"DSD is not merely a modified conventional technique with a longer word length and a higher sampling frequency, but rather is a technique that, by performing direct recording and playback without any filter processing, provides an improved degree of “fidelity to the original” beyond that expressed in specifications while at the same time meeting the requirements of stringent specifications. This allows DSD to communicate the performers’ subtle nuances and the ambience of the performance in an extremely natural manner."

This is the more important because we talk about Music reproduction!

If we cannot debate here, I go away...

Musically
Pat
AudioFEEL - "FEEL the Music"
AudioTouch - Innovative Audio Products
https://www.facebook.com/pages/Audio...01257726677969
Old 7th May 2014
  #4750
Airwindows
 
chrisj's Avatar
Quote:
Originally Posted by AudioTouch View Post
other citation:
"Infinite resolution of the bitstream: In analogue modulators the bitstream is sampled, but not quantized. It still has an infinite resolution just like the analogue signal has. It is superimposed by noise of course, but the more noise you filter out (the lower the low pass filter's frequency limit is), the more of the original resolution is usable - (theoretically) without limitation. "
This is not true. With bitstream as you put it (let's assume one bit at insanely high sample rates, played back without editing) here's how it works…

It IS quantized. It's quantized to something like 'output is more positive than the last sample' or 'output is more negative than the last sample', but that's relative to something. From reading up on delta-sigma integrators it looks like they store voltages in an integrator (like a capacitor?) that drains at a given speed, and record pulses when the integrator's not keeping up with the signal voltage.

To turn this to a longer word length is basically like making a calculation, 'one pulse out of ten cycles means our signal level is 1/10th of full scale'.

But that does not mean DSD is infinite resolution! It's 2 megahertz resolution, or whatever you use. There's a clock involved and that's capturing ONE bit at regular intervals, and that is the digital resolution right there. We can convert it into other things, we could turn it into 256 bit audio at what, 0.001 hz? but that's no different than using simpler means to play the stuff back directly (SACD) and getting a lot of noise and garbage in the high treble, but increasing accuracy as frequency drops.

That's all it is, it's a digital resolution of known data size in which the accuracy increases as frequency drops. We can turn that into 'the same accuracy across the audio band' (various forms of PCM) in known ways, but we're not losing resolution, we're rearranging it.

And it's possible to write dither/noiseshaping to make PCM act like DSD in its output behavior, because I've done that too (and it wasn't that great, but I wasn't targeting 24/192K at the time…)

Do you WANT exactly the same sound as SACD, out of Pono using PCM audio files? This would not be hard to do. It's even set up to roll with that strategy, near as I can tell.
Old 7th May 2014
  #4751
Lives for gear
 
bogosort's Avatar
Quote:
Originally Posted by Sounds Great View Post
No, the question is does it sound better
Did you download Ayre's DSD and PCM files to compare? Might be impossible to do so blindly, unfortunately.
Old 7th May 2014
  #4752
Gear Guru
 
Sounds Great's Avatar
 

Quote:
Originally Posted by bogosort View Post
Did you download Ayre's DSD and PCM files to compare? Might be impossible to do so blindly, unfortunately.
How do I download a DSD file on my computer? And if I can, what do I do with it next?

I have no DSD software or anything that could play such a file, though I do have a couple of SACD players. I've owned one since 2001.
Old 7th May 2014
  #4753
Lives for gear
 
bogosort's Avatar
Quote:
Originally Posted by Sounds Great View Post
How do I download a DSD file on my computer? And if I can, what do I do with it next?

I have no DSD software or anything that could play such a file, though I do have a couple of SACD players. I've owned one since 2001.
Apparently some SACD players will play DSD burned to DVD:
SACD engineers: here's how to make an SACD-R for home testing BEFORE glass mastering!

Or you can buy one of Ayre's PCM+DSD DACs.
Old 7th May 2014
  #4754
Lives for gear
 
nuthinupmysleeve's Avatar
 

Quote:
Originally Posted by Sounds Great View Post
These are nothing more than a continuation of the same discussions we are having here, with the same bias.

--









Unless they're not.



Unless they're not.



Uh, ok. Kind of like records, or tape? So this has what to do with the price of tea in China?



Ya think?



Bingo. So, the big question is...
"most... most... most... most..."

You: "Unless they are not."

???

Exceptions exist. Why do you think the author said "most?"
Old 7th May 2014
  #4755
Gear Guru
 
Sounds Great's Avatar
 

Quote:
Originally Posted by nuthinupmysleeve View Post
"most... most... most... most..."

You: "Unless they are not."

???

Exceptions exist. Why do you think the author said "most?"
I have plenty of SACD's that sound like they came off the same masters as the vinyl. Masters FROM THE 70'S!

I could be wrong, of course, but I don't think so.
Old 7th May 2014
  #4756
Lives for gear
 
nuthinupmysleeve's Avatar
 

Quote:
Originally Posted by Sounds Great View Post
I have plenty of SACD's that sound like they came off the same masters as the vinyl. Masters FROM THE 70'S!

I could be wrong, of course, but I don't think so.
I agree, there are lots of older recordings that could very well be direct transfers from analog to DSD. There are also jazz recordings and classical recordings direct to DSD without digital mastering.

However... "most" = "most"... if you're trying to prove it's inaccurate you need to disprove "most," not provide evidence of exceptions that nobody needs.
Old 7th May 2014
  #4757
Lives for gear
 
nuthinupmysleeve's Avatar
 

Quote:
Originally Posted by TonyVegas View Post
Are you guys seriously still talking about this?
... are you still posting here?

Old 7th May 2014
  #4758
Gear Addict
 

Quote:
Originally Posted by Sounds Great View Post
How do I download a DSD file on my computer? And if I can, what do I do with it next?

I have no DSD software or anything that could play such a file, though I do have a couple of SACD players. I've owned one since 2001.
Download the test files:
Ayre Acoustics Design Thoughts
Download JRiver player:
JRiver Media Center software
Buy a DSD DAC or maybe an astell&kern ak120

I think it would be nice if the Pono did DSD playback.
Old 7th May 2014
  #4759
Gear Guru
 
Sounds Great's Avatar
 

Quote:
Originally Posted by nuthinupmysleeve View Post
I agree, there are lots of older recordings that could very well be direct transfers from analog to DSD. There are also jazz recordings and classical recordings direct to DSD without digital mastering.

However... "most" = "most"... if you're trying to prove it's inaccurate you need to disprove "most," not provide evidence of exceptions that nobody needs.
Does that not miss the point? The media has capabilities that can be utilized, or not! And of course a PCM master is not going to benefit as much (if at all) by getting released on SACD. But proper transfers from master tapes are stunning on SACD's, and nearly as good as (proper) vinyl without the added problems and challenges vinyl brings.

Go and listen to The Rolling Stones 'Let It Bleed' on SACD through your favorite headphones or speakers. And then tell me how much improvement over the CD you hear, even when factoring in any potential confirmation bias. Give it a huge handicap, it still wins. This is NOT like going from 44.1 to whatever with multibit-PCM.
Old 7th May 2014
  #4760
Gear Guru
 
Sounds Great's Avatar
 

Quote:
Originally Posted by walter88 View Post
Download the test files:
Ayre Acoustics Design Thoughts
Download JRiver player:
JRiver Media Center software
Buy a DSD DAC or maybe an astell&kern ak120

I think it would be nice if the Pono did DSD playback, but at $400 wouldn't that be the cheapest DSD player out there?
I have a great Denon player I bought on Craigslist for about 100 bucks. I've got a single disk Sony as well as a 5 disk player that were both under $200 new, some years ago.

I also had a Blue Ray player, forgot which one that played SACD's and had all the hook ups for multichannel as well, I bought it on Craigslist for $40! I sold it because I bought a great Panasonic Blue Ray player for the TV, but it doesn't play SACD's.
Old 7th May 2014
  #4761
Lives for gear
 
nuthinupmysleeve's Avatar
 

Quote:
Originally Posted by Sounds Great View Post
Does that not miss the point?
The point was, like a vinyl pressing, DSD most of the time is a mixdown from PCM or at least is processed/mastered in PCM.
Old 7th May 2014
  #4762
Gear Addict
 

Quote:
Originally Posted by Sounds Great View Post
I have a great Denon player I bought on Craigslist for about 100 bucks. I've got a single disk Sony as well as a 5 disk player that were both under $200 new, some years ago.

I also had a Blue Ray player, forgot which one that played SACD's and had all the hook ups for multichannel as well, I bought it on Craigslist for $40! I sold it because I bought a great Panasonic Blue Ray player for the TV, but it doesn't play SACD's.
Yeah I've got a Pioneer DVD/SACD/DVD-A that I think was $150. If you can burn the DSD files to DVD as in Bogosort's link and you have one of those players listed...... but that's all pretty specific.

Other than that, the current DSD DACs that can play DSF or DFF files are not inexpensive. Seems SACD playback was relatively inexpensive for the consumer. But doing it now with just the DSD files has become relatively expensive for the consumer.
Old 7th May 2014
  #4763
j_j
Lives for gear
Quote:
Originally Posted by chrisj View Post
Yes, absolutely—if I'm not mistaken, it's the feedback that constitutes noise shaping? I know I can make things that act like dithers with or without properly random noise (with different balances of artifacts and noise colors, much like UV22) or things that act like noise shapers (my floating point 'dither' which is only a noise shaper as you can't set a noise level against a floating point wordlength reduction with a varying mantissa).

The trouble is, this stuff so rapidly gets impenetrable to casual bystanders. Is the idea of a 'spectrum from limited rigid specifying of a signal output voltage, to ambiguous representation by averaging out a lot of more fleeting impressions' useful? I'm just trying to explain why DSD is part of the PCM digital audio continuum. It's easiest to call what it does, 'dither', in that it uses statistical density of many brief impressions to generate what seems like a stable output, just as PCM dither suppresses unwanted digital artifacts.

'Cos the thing is you can make the one act like the other. Back in the day I did a noise shaper, 'Ten Nines', that gave something like -160 dB noise floor at 20 hz (while generating an outlandish amount of noise around 20K) from 16 bit 44.1K PCM. All these things can be manipulated if you try. Ten Nines 'dither' would probably sound amazing on Ponos…
You can't do EXACTLY the same thing, and when you look at the noise level in a narrow band you need to remember that you're only seeing part of the noise, of course.

Ultrasonic dither doesn't do what most people think it does, and it is different than noise shaping. This is not saying it's necessarily a bad thing, only that it does something rather different.
Old 8th May 2014
  #4764
Lives for gear
I just saw something that I wish the gearslutz community would consider a little more in these conversations about human hearing and audio gear. Freeman Dyson (a fairly serious scientist) says this:

The whole point of science is that most of it is uncertain. That's why science is exciting--because we don't know. Science is all about things we don't understand. The public, of course, imagines science is just a set of facts. But it's not. Science is a process of exploring, which is always partial. We explore, and we find out things that we understand. We find out things we thought we understood were wrong. That's how it makes progress.
Old 8th May 2014
  #4765
j_j
Lives for gear
Quote:
Originally Posted by MarsBot View Post
I just saw something that I wish the gearslutz community would consider a little more in these conversations about human hearing and audio gear. Freeman Dyson (a fairly serious scientist) says this:
There are many things we do not know. There are some we do (always with the possibility of refinement). There are many things we can disprove (ditto).

This does not explain either Pono or DSD, nor does it provide any kind of excuse for much of the mistaken claims here, which are in either the "can not be demonstrated" or "can not be tested' catagory. If you can't demonstrate something, you can't test it, so such claims are not within the realm of science, and must lie within the realm of personal faith. Personal faith can not be more than personal, no matter how many people share such personal belief.
Old 8th May 2014
  #4766
Lives for gear
 

Quote:
Originally Posted by MarsBot View Post
The whole point of science is that most of it is uncertain. That's why science is exciting--because we don't know. Science is all about things we don't understand. The public, of course, imagines science is just a set of facts. But it's not. Science is a process of exploring, which is always partial. We explore, and we find out things that we understand. We find out things we thought we understood were wrong. That's how it makes progress.

Scio me nihil scire.
Old 8th May 2014
  #4767
Quote:
Originally Posted by j_j View Post
There are many things we do not know. There are some we do (always with the possibility of refinement). There are many things we can disprove (ditto).

This does not explain either Pono or DSD, nor does it provide any kind of excuse for much of the mistaken claims here, which are in either the "can not be demonstrated" or "can not be tested' catagory. If you can't demonstrate something, you can't test it, so such claims are not within the realm of science, and must lie within the realm of personal faith. Personal faith can not be more than personal, no matter how many people share such personal belief.
I think that's all reasonably put.
Old 8th May 2014
  #4768
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laser's Avatar
 

Quote:
Originally Posted by j_j View Post
Is there a point to your interjection, or do you just want to have an argument?

(Sorry, Monty Python seems appropriate here.)
umm....I think the point is: the horse has been long deceased, tenderized, canned, labelled as Alpo, shipped to Pet Smart, devoured by the dog, digested and is stinkin' up the neighbor's rose bushes.
Old 8th May 2014
  #4769
j_j
Lives for gear
Quote:
Originally Posted by laser View Post
umm....I think the point is: the horse has been long deceased, tenderized, canned, labelled as Alpo, shipped to Pet Smart, devoured by the dog, digested and is stinkin' up the neighbor's rose bushes.
Hm. Perhaps it's actually fertilized the roses and is now encouraging flowers, then. :p
Old 8th May 2014
  #4770
Gear Addict
 

Hi all,

I will admit I haven't read the whole topic. I keep glancing at it, it keeps me interested, amused and incredulous all at the same time. Because I haven't been following it religiously, I have a question (or maybe some considerations) which have already been asked and answered, so I beg your indulgence.

To my understanding, Pono is playing back PCM. An Antelope Eclipse (for example) is also playing back PCM. The Eclipse has a dynamic range of 123dB, and 24bit dynamic range is 144dB. Can someone really produce a consumer grade appliance that outperforms a mastering convertor? Simple economic considerations suggest to me that a battery powered device of small capacity will not have the power to outperform a high end converter running on mains

Similarly, can a pocket device power an audiophile headphone amp and match the impedance of any given set of headphones at that price point? Will Pono actually have an analog interface that is sufficient to wow someone who has spent more than 2K on a set of headphones? And will the device be impedance "agnostic" or will it work okay will some headphones, but not with others?

Thirdly, why is 24 bit even necessary in a final file, when to hear the noise floor, the peaks at full resolution would be above the threshold of pain? Why would 24 bit be necessary when a final master on analog tape only has a dynamic range of maybe 110dB if noise reduction is used? How many people are listening in an environment where they can listen to the full dynamic range over the background noise floor (of amplifiers, DAs, breathing, traffic noise etc) without causing themselves permanent hearing damage? How does an amplifier on a mobile device get enough power to deliver that dynamic range from a battery?

Fourth, what vintage microphones capture extended ultrasonics, what ultrasonic components are present in analog recordings that don't cause intermodulation distortions; and what ultrasonics are valuable in any case? Any acoustical vibrations already interact and modulate within the air, so the audible effects of ultrasonics are already present in the audible spectrum, regardless of whether acoustic, analog or digital. Any further ultrasonic information is a degradation of the signal after the microphone has captured it, or even after the final mix has been approved, and is used to define the quality of the playback: by that I mean that intermodulation distortions are considered a flaw in gear design.

If the extended sampling rate is there to"uncramp" the filter, well, there is science to suggest current DA's have issues at such high sampling rates, that could make the accuracy worse, so that the gains of a gentle filter are counterproductive. I will leave that kind of discussion to those who know more technically.

It seems to me that if Pono had developed such technology, they should be making convertors or mic pre's and other amplifiers, because all the music that has been recorded so far has been recorded on mic pre's with worse noise figures, with brownian motion interfering with the glory of 24bit sound. They could be selling their convertors to audiophiles and mastering engineers.

If Pono outperforms an Antelope Eclipse (I only use that as an example as it is expensive and can playback 192/24 PCM files, no other reason) subjectively, the only possible differences I can imagine is how it reconstructs the analog waveform. There are people far more knowledgable than myself talking about this. If Pono has discovered a better set of filters that offer greater accuracy, then why aren't they licensing/selling that tech to the current audio market, to all the AD/DA designers? There would be more return for their investment by fixing the problem with how the music is encoded and decoded in the first place, as that kind of tech isn't a physical asset. If it is simply a dynamic spectrum filter whatchamacallit applied during playback, it is just the next generation of soundcheck in iTunes, distorting the playback in what is currently perceived as more aesthetic.

I am still convinced for myself that the current technology we have is poorly implemented in consumer devices - Pono doesn't have a high bar to clear when the competition is an iPhone. And if they have discovered a great amplifier that can deliver a dynamic range equal to 24bit, I will buy one, and hack it up to make a mic pre, and retro fit the DA's into my converters.
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