I don't expect a modern software interpolation to do the exact same calculations as an SAA7220. The SAA7220 is from the mid 1980's, when the IC processes had much greater feature lengths than now and as a result, digital filtering was relatively expensive. Chances are the designers cut all corners they could cut to keep costs down, like using as short a filter as they could get away with, as coarse rounding of intermediate results as possible and no dither.
Lampie, a few questions not related to your power supply design.
1) Just to confirm, your DAC is dual-mono STARTING FROM the SPDIF input. So, each mono channel has it's own DIR chip as well as it's own DAC chip?
2) How do you then demultiplex each mono-DAC's DIR chip I2S output so to synchronize that only the left or right channel data gets converted by it's DAC chip, ensuring that each mono-DAC receives only mono channel data?
Does your SPDIF source have the ability to selectively transmit only left, or right channel data within the normal L/R multiplexed I2S data frame. So, instead of L/R data framing, maybe it provides L/L data framing to the left channel DAC, and R/R data framing to the right channel DAC?
3) Do you utilize a signal transformer to couple the DAC to the pre-amp, or instead, utilize a differential-amplifier at the pre-amp's input to receive the DAC's balanced output signal?
It is true that all data is seperated already in the transport (Left/Right).
So, yes i use 2 DIR's and then correct the timing plus the phase shift to get a balanced signal out of at least 2 converters per channel. Please be aware that each converter can only handle 192Khz bit rate, so if you like to double this it needs 2 converters to do this (and still have a NOS DAC).
How i do this i also do not want to share at this point.
No transformers in the the analog signal path ! I use some RF filtering here and there.
So, yes i use 2 DIR's and then correct the timing plus the phase shift to get a balanced signal out of at least 2 converters per channel. Please be aware that each converter can only handle 192Khz bit rate, so if you like to double this it needs 2 converters to do this (and still have a NOS DAC).
How i do this i also do not want to share at this point.
No transformers in the the analog signal path ! I use some RF filtering here and there.
Don,
I just stumbled across some posts we had exchanged regarding this same issue, on Audio Science Review, back in 2016!
For example:
"Hi, Don.
I'm not suggesting that brickwall filtering adds any new information, but rather, that it more accurately reveals the information that is there. Said another way, an absence of brickwall filtering will, according to the sampling throrem, prevent the original waveform from being accurately reconstructed. Band-limiting is an rather explicit requirement of the sampling theorem, and is necessary to accurately reconstruct the original continuous time signal. The D/A conversion process produces it's own set of image spectra having noting to do with the effectiveness of the A/D anti-alias filter. D/A converter output exists as discrete steps until/unless the image spectra are removed, thereby rendering the final output as continuous.
Steep slope SINC function FIR digital filters are easily made to exhibit linear phase simply by utilizing symmetrical coefficients in the filter kernel, as most do. Which, of course, is what produces the now infamous pre and post ringing impulse response of such filters, but their phase response is linear. As far as in-band response droop is concerned, even the technically compromised ubiquitous half-band FIR reconstruction filter (the kind incorporated in most DAC chips) are flat to within a very small fraction of a dB, including a zero order hold EQ function, fully up to 20KHz.
All of that said, despite it's near objective perfection, I'm usually most disappointed by the subjective sound of playback utilizing brickwall digital filtering. While I've yet to read a fully convincing technical argument for why that should be, I do believe we will eventually find the key technical parameter or factor that is adversely affecting human pshycho-acoustical perception of sharply band-limited audio sampling and reconstruction, despite the acknowledged mathematical perfection of the sampling theorem.
My nanocent."
I had totally forgotten about that thread.
I'm still spinning my wheels on this question, as you can see. 
I do feel closer to an answer now, though.
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Right, so your NOS DACs are as unsuitable for playing CDs and other physical carriers as most oversampling DACs are.
I don't notice any issues with the CDs I play. I'm too lazy to design in an analog volume control and use the digital volume on my PC which is typically set at -6dB.
On reflection I'd say my designs are more suited to playing material with digital overs as a couple of resistor value changes can be made to alleviate any analog overs. Whereas those DACs sporting digital filters lacking headroom can only be sorted by applying digital attenuation prior to the filter.
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Hi, Lampie,
I don't know whether this is what you've done in your DAC, but the way that I would do this is to time-interleave the two converters, connecting their current outputs directly together so that they sum. Enabling the converter pair to produce double the effective output sample rate which either converter could produce alone. This achieves linear-interpolation entirely within the analog domain. Linear-interpolation leaves the image-bands largely intact, which is no problem in this application because intact image-bands are a feature, not a problem, when it comes to NOS. This time-interleaving technique would be rather simple to achieve in hardware, having to interleave just two converters.
This approach has long been utilized, though. The earliest I know of was by Krell decades ago, in one of their top of the line CD players. More recently, also by Trinity Audio of Germany in their DAC. Also, by ECDesigns in an old DIY project located here at diyAudio.
This does not work with all dual channel converters so single converters are needed then. In this case it becomes more complex as 16bit single ladder converters are not produced (by Philips that is).
BTW, i do not care what other company's do or have done (except from learning from their efforts). I only know what my approach has brought me and i am happy with the results (incl. anyone who has audited these).
BTW, i do not care what other company's do or have done (except from learning from their efforts). I only know what my approach has brought me and i am happy with the results (incl. anyone who has audited these).
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Objective way to figure out something sounding like the real thing or not is to compare the live vs recorded sound side by side with your ears at where the recording microphone was. Any other comparison method is just subjective fluff.
I had this discussion in an other forum and this is my answer:
My local High End audio dealer has organized some life recording sessions for a small number of customers at the broadcast studio in Hilversum (main studios in the Netherlands). Here it is explained how and why certain precautions are made (and compromises). Afterwards the recording was finished we then could compare (as far as we still can remember the life sessions).
Naturally it sounded different then expected as the room acoustics getting into a microphone is different then when sitting on a chair somewhere in that same studio. So even when all is set up according to our high end standard and how we want it to sound thrue the speakers at home it needs a lot of considderations and mastering to get it this far.
So then the next question arrises: How much time we can spend setting up a recording to have it "perfect" before the musicians run away?
Michael Moore & Paul Berner plays "Home". All Paul Berner albums 20% off at SOUNDLIAISON.COM - YouTube
Some recording info:
Recorded in Studio Eleven (Hilversum) with a live audience on November 25, 2012.
This recording is specially made by Sound Liaison for and with lovers of high-end audio recordings
Special for those who own high-end audio equipment will have maximum benefit.
What made this record different is that the file is a one to one copy of the master file (96kHz/24bit)
No quality degradation whatsoever. Enjoy!
Used equipment:
Microphones:
Paul: JZ V67
Michael: JZ V67
Ed: Neumann KM84
Peter: Neumann KM84
Main system - Audio Technica 4022 (AB)
Micpre's: RME Micstacy (Analog > MADI)
Microphone cables: Grimm Audio TPR
Master clock: Grimm Audio CC1
Power Conditioner: Shunyata Research
Mixing headphones: AKG 702 / Sennheiser HD800
Mixing speakers: Grimm Audio LS1
My local High End audio dealer has organized some life recording sessions for a small number of customers at the broadcast studio in Hilversum (main studios in the Netherlands). Here it is explained how and why certain precautions are made (and compromises). Afterwards the recording was finished we then could compare (as far as we still can remember the life sessions).
Naturally it sounded different then expected as the room acoustics getting into a microphone is different then when sitting on a chair somewhere in that same studio. So even when all is set up according to our high end standard and how we want it to sound thrue the speakers at home it needs a lot of considderations and mastering to get it this far.
So then the next question arrises: How much time we can spend setting up a recording to have it "perfect" before the musicians run away?
Michael Moore & Paul Berner plays "Home". All Paul Berner albums 20% off at SOUNDLIAISON.COM - YouTube
Some recording info:
Recorded in Studio Eleven (Hilversum) with a live audience on November 25, 2012.
This recording is specially made by Sound Liaison for and with lovers of high-end audio recordings
Special for those who own high-end audio equipment will have maximum benefit.
What made this record different is that the file is a one to one copy of the master file (96kHz/24bit)
No quality degradation whatsoever. Enjoy!
Used equipment:
Microphones:
Paul: JZ V67
Michael: JZ V67
Ed: Neumann KM84
Peter: Neumann KM84
Main system - Audio Technica 4022 (AB)
Micpre's: RME Micstacy (Analog > MADI)
Microphone cables: Grimm Audio TPR
Master clock: Grimm Audio CC1
Power Conditioner: Shunyata Research
Mixing headphones: AKG 702 / Sennheiser HD800
Mixing speakers: Grimm Audio LS1
My answer was not on a personal level. I do not feel offended in any way.
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Hi, Evenharmonics,
I concur, that test informs one of how closely the recording sounds like the original live event. Certainly, that is the holy grail in electronic playback. In the meanwhile, however, I suggest, that it is nearly as valuable that playback fools your ear-brain in to believing some live event is occurring, and not necessarily the specific live event which was recorded. No, I'm not suggesting this is preferable to the in-studio test, just nearly as valuable an playback characteristic, because we don't have any way of verifying what any recorded live event exactly sounded like that we were not part of anyhow. Which means, 99.9999% of them.
So, we currently rely on engineering specifications as a proxy to guide us as to what an original event sounded like. Since, we have no practical way of verifying this at home. So, at least for me, a playback which sounds live holds almost as much value as whether or not it also sounds like the original event. When something strikes me as live sounding, such as the voice of someone in the house, versus the voice of somone on the TV, my brain doesn't have to consider what I'm hearing, nor rationally weigh the engineering specs. of the playback system. It simply tells me, instantly, that what I'm hearing is live, whether or not it sounds exactly like the original event. Of course, by definition, if playback sounds exactly like the recorded live event, it will also sound live, however that doesn't necessarily mean it would be a more enjoyable musical experience.
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Most if not all preamp, amp and cable companies provide specs. It's not hard to figure out if those devices will provide audibly transparent sound by looking at the measured performance. What's not easy to figure out is the speakers and that's why listening before buying is important for this audio component.
The term "hi-fi" in playback sound means the level of faithfulness of reproduced sound to its input. If someone wants what sounds faithful to his / her preference, then such system should be called "self-fi".
Interesting topic, as i do not want most life experiences in my listening room as most of the time i go out to a concert i have not the spot i would like to have plus in case of a rock concert (i went to a life concert of The Prodogy) it is no fun at all having this kind of "noise" in your house (How about that for your distortion numbers !!!) . So let's be happy that the mic's in a concert hall are positioned above the musicians (always first row)...
Don't you find it interesting that, supposedly, transparent playback rarely sounds anything like an live event when the original recorded performances are live?
Appreciation of excellent technical specifications is intellectual, while appreciation of excellent music is emotional. We should ask ourselves, is our primary listening goal intellectual satisfaction, or emotional satisfaction?
That's certainly one definition. I would expand that, however, to read; "Faithfulness of reproduced sound to the [original acoustic event]." Of course, that raises questions about what constitutes the original acoustic event. And, questions regarding what else we have to guide us aside from set of parameters?
As far as this being 'self-fi; I don't suppose that you listen through anyone else's ears but your own?
Blame the limits of recording / mastering.
I need to ask this to see if we are on the same page, did you mean to talk about replay of musical performance?
We listen to speakers set up in certain location. Unless you know of a way to connect directly from preamp or amp to our eardrums, you can only see the measurements of those components. How well the music album captured the original acoustic event would depend on the skills of recording / mastering engineer. How well preamp or amp performs would depend on how its output is compared to input, which is easy when compared to evaluating speakers and room acoustics.
Non sequitur. Calling component 'X' better than 'Y' because it sounded better to you wouldn't mean anything to someone else who may well have a different taste / preference. Calling component 'X' better than 'Y' because it shows the output that's more faithful to input would mean something to others because the measurements can be shown to them.