What part of “for me” don’t you understand?
Ted, do you have an idea why this vintage DAC chip is still used today compared to more current ones except for money reasons?
It’s a very simple chip. For people who want NOS or R-2R style DACs it’s great. Also most people don’t seem to read the datasheet, the chip requires an opamp to meet its specs (which aren’t great), but many simply use resistors on its outputs which adds a lot of 2nd harmonic distortion and restricts the dynamic range.
If your favorite music doesn’t have a lot of high frequencies and it doesn’t get too loud, the chip works well.
Without an output filter the timing and waveshape of impulses is lost and PRaT will suffer, but if the original recording is already rolled off this won’t be a problem.
I suspect most people that really enjoy orchestral or, say, grunge would be disappointed, but jazz trios, women’s solo voice, etc. should sound fine.
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It’s just a different type of sound and whether or not one prefers it, depends upon personal taste. It’s a nice alternative for those who find the quest for ever more revealing and high resolution to sound a bit stressful. My room is a on the bright side due to windows, thus the relaxed ways of the Border Patrol are a nice recipe. And the tube rectifier is some sort of secret sauce that adds a quality that transcends just the dac chip.
My acceptance of no DSD came when I studied my favorite high quality recordings. Some are PCM, and others are DSD. I realized that a well recorded 16/44 PCM file could sound just as good as a DSD128 file. What mattered most was the quality of the original recordings. For instance, Kenny Burrell’s “Midnight Blue” (Rudy Van Gelder) 16/44 on Tidal sounded just as good as my reference DSD albums. This helped me deprogram the DSD gospel that I’ve adhered to for the last couple of years and welcome the Border Patrol.
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I have a friend who built a PS One based CD player - it sounded very nice, it has to be said, he had a pair of very large full range speakers he had built too, so no passive crossovers “in the way”.
I have an early PS One stashed in the garage should I ever feel the urge to fiddle with such things.
I’m happy with my current kit (with the “accurate” and “relaxed” amp/speaker chains to choose from) but I may get bored in the future…
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Hopefully this won’t cause the audiophile faithful hen house to cluck too much, but I do find certain contradictions in audiophile orthodoxy. For instance, there is a push for lower noise in order to have a pure signal that reveals the truest nature of the original recording.
Once we have the “pure” low noise signal, we then proceed to upsample the bejesus out of it to 20 times DSD. How does that upsampling adhere to the purist ideology? Isn’t upsampling to 20 times DSD a form of addition, of manipulating the signal?
If find a contradiction to all of that. On the one hand, it’s a purist orthodoxy, and on the other we enhance the signal via substantial upsampling.
I’m not a purist. In my world, do whatever one wants to make it sound good. EQ, upsample, downsample, tube stages, computer based software manipulation, spatial enhancers, hire a psychic, I’m fine with all of it. But I do find the “low noise” purism to run contrary to the upsample the heck out of everything philosophy.
Wouldn’t a purist approach be to not over or upsample? Wouldn’t that come the closest to the original recordings qualities?
Cluck away my dear hens!
Come on, man.
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Simple. As Ted has explained many times, higher output sample rates allow the use of a gentler digital filter to remove more high frequency noise.
Note, upsampling is not “manipulating the signal;” all of the information remains exactly as in the original, it is just represented by different numbers.
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Wow, @dancingsea, you have really gone off the deep end here. It’s clear you don’t understand the role noise plays and why upsampling as @tedsmith does in the DSD does nothing but reduce the effect of noise in the audible frequency range. There is no contradiction once you understand what is going on. But hey, you are in the NOS camp now so you won’t see that.
Oh, there is no way in heck that a well recorded 16/44 PCM file could sound just as good as a a well recorded DSD128 file. I have a wonderful SHM-SCAD of “Sticky Fingers” by the Rolling Stones. I also have the SHM-CD that is based off the exact same source. In other words, it is sourced from the same master tapes and utilizes the exact same mastering. Yes, the rips from the SHM-CD sound incredible. But the SHM-SACD rips sounds even better.
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If you compensate your bright acoustics with this DAC, this might make sense for you.
I’d say, everything else you explain is an attempt to justify your choice or question other designs by abstruse technical arguments and assumptions.
But that’s not necessary anyway, if it works for you like that, it’s fine.
I just really didn’t know that 20-30 year old DAC chips have a fan base and why and that they at least seem to be halfway competitive for certain music with certain frequency response limitations or at certain levels.
DAC‘s are made to make their buyers happy, so this one seems to have reached the goal for you and that’s all it needs! I’m sure what you hear in your environment is true there and to your ears.
I’d really be interested how a newly made DAC based on this old chip sounds. My memory on all the DAC‘s I owned before since the one with a similar chip are so bad, that I wondered how this first chip can sound listenable or more than that today, but it seems so.
I’m not trying to justify my choices, just explaining my process.
As for the noise and upsampling thing, there’s for a sure a contradiction. Upsampling does alter things, it is not “pure”. I’m in no way advocating one side or the other, just observing the obvious that a DAC, any DAC, by it’s very nature, and no matter the price tag, is doing some form of altering the signal. It may be for the best, but it’s still an alteration.
Life is full of contradictions. Audiophile land is no different. The curious part is how we adopt a belief which then blinds us to the contradictions.
I’m all for upsampling. But claiming it is more pure is a delusion. It is a manipulation of some sort. My observation is about the delusion, not about being for or against upsampling. In my own experience, I have found there to be pros and cons to the upsampling process.
Let’s pause for a moment of forum psychology. Just because someone offers a different perspective does not mean that your brain has been invaded by marauders and must be either defended against or attacked. It just means someone has a different point of view. And that’s ok.
Simple logic shows that not over or upsampling is less of a manipulation. Whether or not that’s better, is entirely up to you.
multiplying sample rate by an integer is not manipulating the signal.
to say otherwise is ignoring basic maths, which is “flat earth” territory.
as explained carefully and often in this thread. let it go man 
In particular, if the DS is upsampling by 2, then every other sample is identical to the input. The other every other sample contains the exact same information, the samples are filled in assuming the ADC was band limited, which all are.
The neat thing is that if you throw away the original samples in the upsampled stream and then upsample it again you get the original samples as the new fillins. It’s hard to claim that the signal is being manipulated or distorted by proper upsampling. Nothing is lost…
(The DS never upsamples by just a factor of two, but the same thing happens: upsample by 147, pick a sample, skip 146, pick the next, skip 146… and you have the original information. If you then upsample that by 147, you’ll find your exact original samples every 147 samples if you start at the correct point.)
Not using an upsampling filter is what grossly changes the result. Especially in the high frequencies.
Not understanding the principles is the delusion. It’s not a matter of opinion.
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Another way to see that filtering (with upsampling) is good and that not filtering (and not upsampling ) is bad is to realize that the sampling clock is very unlikely to exactly hit the peaks of the music. If you don’t filter, the height and position of the peak between two samples is lost. If you filter correctly (which requires upsampling and an analog filter) you’ll get the correct peak’s height and it’s correct timing… (All assuming the input was bandlimited, which is standard.)
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A critical point in this discussion.
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I just grabbed a few pictures from https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings to make my point (tho the subject of that page isn’t sampling per se, it’s still interesting.) Here’s a sine that someone in their infinite wisdom decided to maximize the sample values:
Note that just by looking at the samples and not using a filter the best you could guess is something like a square wave!
But filtering is just the reverse of the above picture: filtering the digital samples gives the correct audio shown. But such an analog filter can’t be built without oversampling. Here’s the result of oversampling:
(Note that the original samples are still there!) It’s obvious that even without much filtering you are much closer to the original signal. Now it’s possible to build an analog filter to filter the upsampled samples and you’ll get back exactly the original audio.
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Hey @tedsmith, thanks again for taking the time to share your knowledge. I am having trouble grasping the role of filtering in your example. How does a filter help put a sample at the 1.5 point of the scale that is in the picture you used?
A weird way of looking at it is that the analog filter is like an infinite resolution upsample 
Digital upsampling is done by adding zeros between the samples (e.g. for upsampling by 147, you’d add 146 zeros between every sample.) Then you do a (steep slope) low pass filter at (approx) 1/2 the sample frequency - that changes all of the zeros to the samples that would have to been there if the analog input was antialias filtered at 1/2 the sample frequency. The tech is the same for analog as digital, except you can’t do as steep of a filter in analog so you need digital oversampling (with it’s steep filter) and then you have room for a gentler analog filter.
That’s the whole point of the sampling theorm: you can sample to digital and as long as the input was bandlimited to 1/2 the sample frequency you have all of the original information. To get the analog back you filter again at 1/2 the sample frequency. But the trick is that last filter can’t be done in analog without really rolling off the top end or allowing aliasing. With upsampling you can do the steep filter digitaly and then filter again in analog but at a higher sampling frequency so its easier (or even possible.)
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By the way if this was obvious there wouldn’t be soooo much horrible information about it on the net. If you want to understand it you should look at papers and books from universities (you might need at least a calculus background.)
And I’m not putting in all of the caveats needed for the sampling theorem to hold, they would just confuse things a lot at this level. They all have a much smaller effect than the difference caused by not upsampling or not filtering.
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I find NOS DACs difficult to listen to, especially with no external upsampling/filtering. There is something about the sound that rubs my brain the wrong way. I don’t even like it with Jazz trios, much less 60’s and 70’s Rock…