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DSD upsampling over original format


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Thanks very much Jud. That was a great synopsis for someone like me who can be a bit overwhelmed by a lot of the technical speak. I'm going to bookmark this one. As someone who lived through the early days of digital audio (my first CD player was Radio Shack branded model purchased around 1984 when I was in college) I can attest to the unbelievable marketing that went on in the early days. A lot of the music did sound pretty harsh in the beginning and it took a while for our ears to convince our brains of that.

Digital System: Cybershaft 10MHz OCXO clock premium>Antelope Liveclock>RedNet D16>AES Cable>Mutec MC-3+ USB​>AES Cable>Schiit Yggy

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The reason Jud and Miska mention this is that many consider it advantageous to do this upsampling on a PC and not internally in your DAC chip. The idea is twofold: the more powerful PC can do a better job of conversion and also allow you - if you have the proper software - to use different filtering, modulating, and dithering that may make the sound more to your liking than what's built into your DAC chip. The other claimed benefit is that if you do this upsampling to DSD on your computer using software like Miska's (or with JRiver), then you skip that difficult conversion stage inside your resource constrained DAC, which should mean better SQ on the analog end.

 

 

I can see that this might make a lot of sense if you are also using a DAC purpose designed not to upsample (isn't this the theory behind the XX high End/Phasure pairing?). But used in conjunction with an upsampling DAC, what evidence is there that the "difficult conversion stage" in the DAC is actually by-passed? Maybe this approach could actually degrade SQ, involving 2 conversion stages rather than one? For example, I believe I read somewhere that Chord recommend that input to the Hugo should only be at native sample rate for this very reason, as the Hugo upsamples/transcodes everything to ultra high PCM anyway, with I guess the implication that prior upsampling actually makes the DAC's job harder.

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I can see that this might make a lot of sense if you are also using a DAC purpose designed not to upsample (isn't this the theory behind the XX high End/Phasure pairing?). But used in conjunction with an upsampling DAC, what evidence is there that the "difficult conversion stage" in the DAC is actually by-passed? Maybe this approach could actually degrade SQ, involving 2 conversion stages rather than one? For example, I believe I read somewhere that Chord recommend that input to the Hugo should only be at native sample rate for this very reason, as the Hugo upsamples/transcodes everything to ultra high PCM anyway, with I guess the implication that prior upsampling actually makes the DAC's job harder.

 

There is no evidence. It's about which sounds better to you. I think the real advantage is when using a less-than-SOTA DAC, the computer can do the heavy lifting instead, thus elevating the DACs sound quality beyond it's price point. Chord has a proven method of achieving great sound and it seems many (including me) find software upsampling before one of their DACs is not preferred. It's all subjective.

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There is no evidence. It's about which sounds better to you. I think the real advantage is when using a less-than-SOTA DAC, the computer can do the heavy lifting instead, thus elevating the DACs sound quality beyond it's price point. Chord has a proven method of achieving great sound and it seems many (including me) find software upsampling before one of their DACs is not preferred. It's all subjective.

 

Yeah, but two parts to Norton's question and I want to be clear on both.

 

First: DAC chips have to be programmed to only do *necessary* sample rate conversion or else the sigma-delta modulator won't get the proper "8x" sample rate fed to it. Feed it 44.1 and the bitstream will be run through three rounds of doubling until it gets to 352.8. 88.2 input gets doubled twice, 176.4 once, 352.8 (if the DAC accepts it) not at all. DSD input bypasses this PCM interpolation step entirely.

 

Whether sample rate conversion before the DAC results in better sound is something about which various objective measurements can be done but is ultimately a matter of taste.

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 -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Yeah, but two parts to Norton's question and I want to be clear on both.

 

First: DAC chips have to be programmed to only do *necessary* sample rate conversion or else the sigma-delta modulator won't get the proper "8x" sample rate fed to it. Feed it 44.1 and the bitstream will be run through three rounds of doubling until it gets to 352.8. 88.2 input gets doubled twice, 176.4 once, 352.8 (if the DAC accepts it) not at all. DSD input bypasses this PCM interpolation step entirely.

 

Whether sample rate conversion before the DAC results in better sound is something about which various objective measurements can be done but is ultimately a matter of taste.

 

Thanks Jud. I didn't even want to try to explain (which you do succinctly).

 

There are endless and tiring arguments here and elsewhere regarding "what's better/best" and I think "ultimately a matter of taste" is often forgotten but should be the take-away at this level of our hobby.

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Filters?

 

OK, digital audio is based on the Sampling Theorem (the name Nyquist gets mentioned a lot, though there were other guys too, like Whittaker and Shannon). Basically what the Sampling Theorem says is, in the world of mathematics, you can take samples of any signal (like music) and reconstruct it to mathematical perfection, as long as you take samples more than twice as fast as the signal changes (more than twice the highest "frequency of interest," which with music meant for us humans is the upper frequency limit of our hearing, ~20,000Hz). Thus the CD sample rate of 44,100Hz, a little more than double 20,000Hz. Once you've got the samples, you filter out the higher frequency half (the higher frequencies are stopped, the lower frequencies pass, so it's called a "low pass" filter), and you're left with the lower frequency stuff - music!

 

What I don't understand from this is why you need to filter out the "higher frequency half" if it 's inaudible - is this the thinking behind NOS DACs?

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What I don't understand from this is why you need to filter out the "higher frequency half" if it 's inaudible - is this the thinking behind NOS DACs?

 

Terrific question. Two answers:

 

- See the part of the long post above that talks about the audible aliasing (mirroring) distortion that takes place when there's leakage of ultrasonic frequencies through a filter. With no filter at all, there's not just leakage, there's a flood. No doubt you'd hear music, but with lots of audible distortion.

 

- NOS DACs aren't filterless; they're like the early CD players, with just the anti-aliasing filter, no interpolation or interpolation filters. Either you feed them a bitstream that's already been interpolated in software, or you get aliasing distortion from leakage through the filter. Some people prefer these DACs with RedBook because the distortion subjectively gives an exciting, "hot" sound.

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 -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Hot sound? Maybe a natural & organic sound. Taste. Yeah.

 

If you listen to a favorite track in RedBook and higher resolution (different masterings can be taken out of the equation by offline converters), do they sound at all different to you, and if so, how?

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 -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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If you listen to a favorite track in RedBook and higher resolution (different masterings can be taken out of the equation by offline converters), do they sound at all different to you, and if so, how?

 

Yes, of course they do but what is your point Jud? Are your measurements subjectively more pleasing to my ear? As stated previously, we've gone around this mulberry bush countless times. Better? Best? It's a matter of taste ultimately.

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Yes, of course they do but what is your point Jud? Are your measurements subjectively more pleasing to my ear? As stated previously, we've gone around this mulberry bush countless times. Better? Best? It's a matter of taste ultimately.

 

No inquisition, I was just curious how you would describe the differences. You said "natural" for the sound of RedBook; I wondered if higher res sounds - what? Accurate? Sterile? Both? Or something else?

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 -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

Link to comment
Terrific question. Two answers:

 

- See the part of the long post above that talks about the audible aliasing (mirroring) distortion that takes place when there's leakage of ultrasonic frequencies through a filter. With no filter at all, there's not just leakage, there's a flood. No doubt you'd hear music, but with lots of audible distortion.

 

- NOS DACs aren't filterless; they're like the early CD players, with just the anti-aliasing filter, no interpolation or interpolation filters. Either you feed them a bitstream that's already been interpolated in software, or you get aliasing distortion from leakage through the filter. Some people prefer these DACs with RedBook because the distortion subjectively gives an exciting, "hot" sound.

 

Thanks. I want my next steps in digital audio to be more guided by an understanding of how things work, rather than just by product marketing and reviews.

This is very helpful.

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