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Modern Multi-Bit DAC vs Delta Sigma, specifically AKM's newest flagship, but also others

@mocenigo: I thought the PCM1791A used a version of van de Plassche's scheme that "rotated" among current sources (sinks)? Don't remember now, and am no longer sure how to reach the guys I knew at BB before TI took them over...

Segmentation, using unary (unit-weighted) cells for the MSBs and binary weighting for the lower bits, is very common in R-2R DACs. It is impractical to match (even with trimming) the MSBs otherwise (although other schemes using digital compensation are also popular).

Many devices have used charge pumps but they do have their own problems, including limited current, relatively high supply output impedance (various schemes to fix that are used), and limited current capacity. SMPS schemes are much more widely used these days.

This thread briefly discusses segmented DACs: https://www.audiosciencereview.com/...ital-audio-converters-dacs-fundamentals.1927/ An n-bit DAC requires roughly 2^-N matching among MSB cells, something very difficult to achieve for high-accuracy DACs. Segmented designs are one way, delta-sigma modulators another.
 
What does 4 level delta sigma modulation for the least significant bits mean?
I just looked through the 1791A's datasheet and didn't find any info alluding to its nature - how did you find out about this? And the resistor array?

Look at the 1791A data sheet
page 52. The same description is also in the 1793 data sheet.

You seem to know about the inner workings of DAC chips

Not really, I have read some introductory chapters in books about DACs, and some papers, and I understand the mathematics. I also have read a few data sgeets.
- have you ever come across a designs that used a charge pump to increase the source voltage for the most significant bits to increase accuracy?

Intuitively this risks to be quirte imprecise. The gain would have to be absolutely perfect.

Or one that uses extremely HF PWM to regulate the current of the least significant bits, which use the next LSB that's not PWM'd as reference to operate (set as a fraction of the reference)?
If you've never seen either thing implemented, I think they'd both be pretty cool to see in a design - as long as there's nothing inherently wrong with either lol

I think that the 1791/2/3 does something similar.
 
@mocenigo: I thought the PCM1791A used a version of van de Plassche's scheme that "rotated" among current sources (sinks)? Don't remember now, and am no longer sure how to reach the guys I knew at BB before TI took them over...

It is a bit more complex than the scheme I presented. The 6 MSBs are turned into ICOB, and the LSB are joined with the MSB and modulation is applied to that and reduced to 5 levels (not 4, i was wrongo). So the first set of values from 0 to 62 and the second one from 0 to 4 are added. Ao you have to represent values from 0 to 66, and this is passed through DWA. It may be rotating or randomising in some other way, so I do not know how DWA is performed.

[...]

This thread briefly discusses segmented DACs: https://www.audiosciencereview.com/...ital-audio-converters-dacs-fundamentals.1927/ An n-bit DAC requires roughly 2^-N matching among MSB cells, something very difficult to achieve for high-accuracy DACs. Segmented designs are one way, delta-sigma modulators another.

Thanks!
 
It is a bit more complex than the scheme I presented. The 6 MSBs are turned into ICOB, and the LSB are joined with the MSB and modulation is applied to that and reduced to 5 levels (not 4, i was wrongo). So the first set of values from 0 to 62 and the second one from 0 to 4 are added. Ao you have to represent values from 0 to 66, and this is passed through DWA. It may be rotating or randomising in some other way, so I do not know how DWA is performed.
Had to look it up again, wasn't sure what "ICOB" and "DWA" were (senility, and everyone invents their own TLAs). I see it does use a DS section for the lsbs (not the MSBs, forgot about that). IIRC the design was at least partly to reduce the oversampling ratio required for 24 bits by using a more conventional MSB segment with a DS lsb "segment". The block diagram is a bit confusing, probably intentionally. DWA is in the digital domain. Old design, still good.
NP; that was more for others since you seem to understand it already.
 
Look at the 1791A data sheet
page 52. The same description is also in the 1793 data sheet.
Cool DAC.
Intuitively this risks to be quirte imprecise. The gain would have to be absolutely perfect.
Before you replied, DonH56 saw and also responded, interestingly he thinks he's seen an implementation
I think that the 1791/2/3 does something similar.
the 1791 didn't mention in its "theory of operation" section you linked me to - are the 2/3 are different? I think it'd probably be a bit redundant for PWM + DWA on a 5 bit sigma delta DAC, but that's just my initial assumption
 
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Cool DAC.

Before you replied, DonH56 saw and also responded, interestingly he thinks he's seen an implementation

the 1791 didn't mention in its "theory of operation" section you linked me to - are the 2/3 are different? I think it'd probably be a bit redundant for PWM + DWA on a 5 bit sigma delta DAC, but that's just my initial assumption

Page 53, not 52, of the linked document. It is described there.
 
To the mod who moved this OP to a massive thread about DAC sound signatures (and everyone else reading because I added more info to the thread in here too): this thread is not about sound signature, it's about the internal operation of recently manufactured multi-bit DAC chips (specifically the new AK4499 Velvet, but also others), which I've come to understand may be imposters compared to others. Imposters? Imposters! :not true multi-bit chips like the ones made in the past, but somehow a hybrid of delta sigma + multi-bit, which, being different, may affect performance. At the end of the thread I did have some words on the sound quality of my L70 Velvet, because to me, it sounded vastly superior to the [too many] other Topping DACs I have. Fun fact: the only reason I currently know the AK4499 Velvet in my E70 Velvet is a multi-bit design, is because I was looking for a possible explanation for my perceived [but maybe imagined..] difference in clarity between it and my DX1, E30 II, E30II lite, and E50 (I do have reasons for all of those except the E50, which I planned to return but missed the window for). ANYWAY! That bit at the end of the thread was very secondary to its primary focus, which, again (to be sure), is not discussing the implementation of the AK4499 Velvet in the Topping L70 Velvet, but the AK4499 Velvet itself: information on its design and basic operation, and maybe information on other, more recently manufactured multi-bit DACs if they're similar. I removed what I had at the end of the OP (regarding sound comparison) so there is no confusion.

Thread:
I've got AKM's 4499EX in my Topping E70 Velvet. I'm under the impression that the 4499EX is a two-chip solution - the 4499EX + comes with a 4191EQ

From AKM: "The AK4191EQ is a new concept Multi-bit stereo Premium Digital Data Converter employing VELVETSOUND™ technology. By using the AK4191EQ to process the digital signal for D/A conversion, we have minimized the effects of digital noise within the analog output, resulting in a perceived improvement of the ratio of signal to noise. The AK4191EQ has a built-in digital filter with multi-bit sigma delta modulator and 256 times oversampling processing. A wide variety of music can playback by inputting 1536kHz PCM data and DSD1024 data."

I understand this to mean the AK4191 is the chip that goes between the digital source (coax, spdif, USB, HDMI etc., etc.) and the AK4499.

What I don't get, though, is why there's a sigma delta modulator in it though... wouldn't that nullify the [alleged] benefit of the AK4499 being multi-bit?

Something else that's confusing is AKM says the 4499 and 4191 are separate to keep the analog filtering and digital processing in separate packages, but then they go ahead and say the 4919EQ processes the digital signal so that there's less noise on the 4499's eventual output... So if the 4191's input is already digital, what's this sigma delta going on for? T

And another thing: the low-pass filter on the analog audio output... you know, the thing that usually brings things down to -60dB to -100dB by 24kHz or so with a 44.1kHz... is that still in the 4499? Or is it in the 4191? So much of what AKM has said appears to be contradictory or just plain impossible!

Also, with the AK4499 Velvet (technically AK4499EX, and when packaged with the 4191EQ, called the AK4499EXEQ):
Top of Page 1 of datasheet for proof:
View attachment 352942
Where is its resistor bank? Is it in the chip???

I'm pretty new to exploring the intricacies of DACs past their advertised specifications and audible sonic qualities, so sorry for if any of the questions are pretty stupid... (Not an excuse but a reason: I had a desktop USB/toslink/coax DAC I was really happy with for almost 10 years, and it broke (accidentally, by my hand :( ), and now I'm interested in learning more about DACs. I do think I've been lucky to find my replacement as quickly as I have, but the journey isn't over yet -I'm still trying to learn more

And finally (& maybe most importantly...) : is the AK4499 a good example of a modern multi-bit DAC - basically made the same way but with slightly newer manufacturing technologies -, or is it an imposter - a delta-sigma DAC masquerading as multi-bit (like I said, I haven't been looking long, but I've briefly come across people saying some modern multi-bit DACs are weird hybrids or something... from what I've been able to extrapolate from what's been suggested here, there, and everywhere: a delta-sigma type DAC with unconventional output is run in a way (maybe at a higher frequency?) that it's able to output like a multi-bit DAC through multiple pins, but there are less pins and resistors and the clock is doubled up (well not doubled, but maybe 10x - it's like 5 bits instead of 20 or 24 bits). I don't know, it's messed up what I've seen and I'm tired of guessing so I'm hoping someone knows what's going on!
This is a very clear and enlightening discussion of the AKM velvet implementation! Thanks.
 
I just don't get the need to use the term "imposter". A "pure" multi-bit DAC is not really desirable.

These modern hybrids are using multiple weighted 1-bit DACs instead of trying to give every single bit of the original information a weighting.

It gives you the benefit of neither having to run a 1-bit DAC at a stupidly high frequency, nor reaching the downsides of weighting more than a handful of bits.

There's really nothing to gain from "purity" in this case.
 
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