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post #1336 of 17528

Jason, if I understand you correctly then todays ADCs are also not bitperfect. Wouldn't a bitperfect DAC only fully make sense if the sound it reproduces has been recorded and processed with bitperfect equipment?

post #1337 of 17528
Thread Starter 
Quote:
Originally Posted by AManAnd88Keys View Post
 

Jason, if I understand you correctly then todays ADCs are also not bitperfect. Wouldn't a bitperfect DAC only fully make sense if the sound it reproduces has been recorded and processed with bitperfect equipment?

Todau, this is true in most cases, but saying, "Hey, the system is screwed, might as well say 'the hell with it'" doesn't move us forward. Plus, there's still plenty of music recorded with multibit ADCs out there. 

post #1338 of 17528
Thanks for the info Jason.

Back in the late 1980s-early 1990s I remember that Mike Moffat teamed up with Nelson Pass to do a custom over top analog and digital recording setup for Mobile Fidelity Sound. Is that still around somewhere?
post #1339 of 17528
Quote:
Originally Posted by Jason Stoddard View Post
 

First cost cut: digital filtering (oversampling). That allowed manufacturers to throw out the analog brick wall filter, which was wwayyyyy cheaper. Unfortunately, all digital filter algorithms (except one) throw out the original samples in the process of upsampling. At that moment, the concept of "bitperfect" went out the window. Ah, well. It was cheaper.

 

Jason, can you tell us more about this closed-form Digital Filter? What are the working principles?

 

Thanks. 

 

Leonel

post #1340 of 17528
Thread Starter 
Quote:
Originally Posted by Jones Bob View Post

Thanks for the info Jason.

Back in the late 1980s-early 1990s I remember that Mike Moffat teamed up with Nelson Pass to do a custom over top analog and digital recording setup for Mobile Fidelity Sound. Is that still around somewhere?

Yep, it is. 2 chassis, about 100 lbs or so of gear, per stereo ADC.

 

Maybe I can talk him into doing a modern version.

 

Don't get excited--if we do it, it will be STUPID expensive. I doubt if we'd replace many studio ADCs with it. It would really be for nutcases.

post #1341 of 17528
Quote:
Originally Posted by Jason Stoddard View Post
 

Todau, this is true most cases, but saying, "Hey, the system is screwed, might as well say 'the hell with it'" doesn't move us forward. Plus, there's still plenty of music recorded with multibit ADCs out there. 


Right, that's why I think your approach is awesome. Many people are used to unnatural sound, because every part of the chain introduces something to the signal that shouldn't be there in the first place. You guys want to reverse that "evolution" (at least partially, bitperfect is a great start), I am really looking forward to the new DAC.

post #1342 of 17528
Quote:
Originally Posted by Jason Stoddard View Post
 

Todau, this is true most cases, but saying, "Hey, the system is screwed, might as well say 'the hell with it'" doesn't move us forward. Plus, there's still plenty of music recorded with multibit ADCs out there. 

 

Jason, are you referring to early Digital recordings or are there still Multibit ADCs being employed today?

 

Thanks.

post #1343 of 17528
Thread Starter 
Quote:
Originally Posted by rocksteady65 View Post
 

 

Jason, can you tell us more about this closed-form Digital Filter? What are the working principles?

 

Thanks. 

 

Leonel

 

Until Mike finishes his white paper, all I can say is what I've said before: it's based on a 1917 Western Electric paper on pulse code modulation, with an algorithm perfected by a Professor Emeritus in Mathematics at Iowa State (to get around the divide-by-zero problem) and implemented by a Rand Corp mathematician, now running on our own DSP with 18000+ filter taps. 

 

Funny convo at TheShow: a reviewer asked Mike "what digital filters will Yggy use?"

 

Mike: "The right one."

 

"No, I mean is it slow roll off, apodizing, etc--"

 

Mike cut him off, "It's the right one."

 

"Well, some people like a choice of different filters--"

 

Mike groaned. "This isn't a fashion show. This isn't a popularity contest. Yggy uses the only closed-form digital filter and retains the original samples."

 

The reviewer seemed confused. "But what about other digital filters--"

 

"This isn't Burger King. This is math."

 

"What if someone likes something else?" the reviewer pressed.

 

"That's fine. We'll continue using the right filter," Mike told him.

post #1344 of 17528
Quote:
Originally Posted by Jason Stoddard View Post

Until Mike finishes his white paper, all I can say is what I've said before: it's based on a 1917 Western Electric paper on pulse code modulation, with an algorithm perfected by a Professor Emeritus in Mathematics at Iowa State (to get around the divide-by-zero problem) and implemented by a Rand Corp mathematician, now running on our own DSP with 18000+ filter taps. 

Funny convo at TheShow: a reviewer asked Mike "what digital filters will Yggy use?"

Mike: "The right one."

"No, I mean is it slow roll off, apodizing, etc--"

Mike cut him off, "It's the right one."

"Well, some people like a choice of different filters--"

Mike groaned. "This isn't a fashion show. This isn't a popularity contest. Yggy uses the only closed-form digital filter and retains the original samples."

The reviewer seemed confused. "But what about other digital filters--"

"This isn't Burger King. This is math."

"What if someone likes something else?" the reviewer pressed.

"That's fine. We'll continue using the right filter," Mike told him.


That is the best post in this thread. Mike is awesome!
post #1345 of 17528
Quote:
Originally Posted by Jason Stoddard View Post
 

 

Until Mike finishes his white paper, all I can say is what I've said before: it's based on a 1917 Western Electric paper on pulse code modulation, with an algorithm perfected by a Professor Emeritus in Mathematics at Iowa State (to get around the divide-by-zero problem) and implemented by a Rand Corp mathematician, now running on our own DSP with 18000+ filter taps. 

 

Funny convo at TheShow: a reviewer asked Mike "what digital filters will Yggy use?"

 

Mike: "The right one."

 

"No, I mean is it slow roll off, apodizing, etc--"

 

Mike cut him off, "It's the right one."

 

"Well, some people like a choice of different filters--"

 

Mike groaned. "This isn't a fashion show. This isn't a popularity contest. Yggy uses the only closed-form digital filter and retains the original samples."

 

The reviewer seemed confused. "But what about other digital filters--"

 

"This isn't Burger King. This is math."

 

"What if someone likes something else?" the reviewer pressed.

 

"That's fine. We'll continue using the right filter," Mike told him.

"This isn't Burger King. This is math."

 

HAHAHAHAHahahahahahahaha…

 

Or maybe B&R 31 flavors, with the ever popular, flavor of the month filter…

 

It just struck me as 'odd' that high end dacs could even HAVE a choice of what is accurate.

Like there is a choice of more than one version of a digital bit stream conversion.

 

And I understand why it's that way and all, but still how can you have multiple choices of a precise and exact re-creation of the original analog signal?

 

Its all FM  (freak'n magic)… :atsmile:

 

JJ

post #1346 of 17528

Oh boy. This does not bode well for my wallet.

post #1347 of 17528
Quote:
Originally Posted by Jason Stoddard View Post
 

 

Until Mike finishes his white paper, all I can say is what I've said before: it's based on a 1917 Western Electric paper on pulse code modulation, with an algorithm perfected by a Professor Emeritus in Mathematics at Iowa State (to get around the divide-by-zero problem) and implemented by a Rand Corp mathematician, now running on our own DSP with 18000+ filter taps. 

 

Funny convo at TheShow: a reviewer asked Mike "what digital filters will Yggy use?"

 

Mike: "The right one."

 

"No, I mean is it slow roll off, apodizing, etc--"

 

Mike cut him off, "It's the right one."

 

"Well, some people like a choice of different filters--"

 

Mike groaned. "This isn't a fashion show. This isn't a popularity contest. Yggy uses the only closed-form digital filter and retains the original samples."

 

The reviewer seemed confused. "But what about other digital filters--"

 

"This isn't Burger King. This is math."

 

"What if someone likes something else?" the reviewer pressed.

 

"That's fine. We'll continue using the right filter," Mike told him.

 

Ha ha! Excellent answers. Refreshingly devoid of BullSchiit! :beerchug:

post #1348 of 17528
It makes me sort of depressed that even most high-end DACs are designed to cut costs and not produce good sound quality. I defend digital sound all the time to non-audiophile friends that think vinyl sounds better (they haven't heard any digital converters but their iPod... hardly a fair fight), so it makes me mad that everyone cuts corners like that.
post #1349 of 17528

wow, that's an incredibly creative rewrite of digital audio history... but not particularly accurate.  Here are some readily verifiable facts:

 

1. 44.1KHz sample rate was chosen well before CDs were invented.  This was during the first era of consumer digital recorders, which recorded on videocasettes.  That rate turns out to allow you to hold 3 stereo samples per video line, and still give you a little more than 2KHz transition band for antialiasing filters.  Redbook adopted 44.1K/16b because sony won the first format war (philips wanted 44.056Khz), and they wanted to leverage all the recordings already made in that format.

 

2. those devices had a choice of 14 or 16 bits as there were two competing formats, from sony and philips.  The philips format was 14bits, which by the way, was still significantly better than any analog tape recorder of the time (in theory).

 

3. before these recorders, digital audio recorders were already being used professionally.  (A popular sample rate back then was 50KHz).  Apogee Electronics got their start back then making high quality antialiasing filters for some of these machines.

 

Most of these used SAR DACs and ADCs, which was really the only way to get >14 bits.  There is no way to match resistors closely enough to do R-2R DACs with that resolution, 

Regardless, both types suffer from large amounts of differential nonlinearity, well over 1LSB, which is much more audible and objectionable than large amounts of integral nonlinearity.

 

Digital oversampling filters were introduced not to cut cost (those chips were quite expensive at the time) but to improve the whole process of antialiasing.  It allowed the freedom for the analog filters to have much more desirable characteristics, like lower Q (less ringing), better headroom, less critical component matching requirements, etc.

 

Sigma-delta converters ICs were introduced in the 80's not to cut cost (the first ones, manufactured by dbx, were very expensive) but to improve audio quality.  They had superior specs and sound, due mostly to the lack of any measurable differential nonlinearity, and very smoothly shaped integral nonlinearity.

 

It turned out that, once the theory of sigma delta converters was better understood by the engineering community, the process to manufacture them could be made far cheaper than high-resolution SAR converters.  But part of that is just the march of technology, as it's possible to get very good, inexpensive SAR converters these days, based on some of the same technological advances that were driven in part by development of S-D converters.

 

So, the cost reduction followed the innovation, but wasn't the impetus for it. These advances were done by engineers dedicated to improving the audio experience (even if they didn't always succeed).

 

The main downside with most current SD converters is they do not have very good accuracy at DC (though there are some around that do), and they have a few mS of latency due to digital filtering which can interfere with industrial uses such as motor control applications where they are in the feedback loop. 

 

Bottom line is, for conversion to/from analog, one needs antialiasing and anti-imaging filters and a sampler/quantizer.  Almost from the beginning, the filtering process has been a combination of analog and digital filtering.  S-D converters are a result of looking at the theory and coming up with a much more elegant solution, though one that because of the heavy duty math involved, isn't exactly intuitive.

 

I'm still scratching my head at what bitperfect is supposed to mean in this context.  I can't come up with any logical explanation.  Analog signals do not have any 'real bits' or 'intrinsic bits' hidden inside them...

post #1350 of 17528
Before it is converted they change the bits using digital filters, etc which results in a different waveform even before it's converted to analog.
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