Which DACs bypass digital filtering?

1 hour ago, mansr said:

Perhaps it's best to study a few examples.

 

Below is a block diagram of the TI DSD1793 chip used in iFi DACs. The PCM1795 in the TEAC UD-501 is similar.

This shows a standard PCM path with 8x upsampling followed by sigma-delta modulation. A filter bypass mode allows direct input to the modulator at 384 kHz. The datasheet reveals that this is in fact a hybrid design where the modulator, a 3rd order 5-level design, operates on the low 18 bits only. The output is combined with the high 6 bits to form a 66-level code which forms the input to the actual D/A conversion stage.

 

The DSD path is separate and does not involve any digital processing. The D/A stage supposedly consists of a shift register style FIR filter similar to Miska's design but with different weights on each position. The datasheet is vague but supports this idea. Four different filter choices are available.

 

Now look at AKM's top range. Several variants with similar design are available. They show up in devices from TEAC (UD-503), Linn, ESOTERIC, Marantz, and others. This diagram is from the datasheet of the AK4497:

Again, a fairly typical PCM path. The main difference compared to the DSD1793 is the addition of a digital attenuator (DATT). A filter bypass mode is available. I can't find any information on the sigma-delta modulator, but I would assume it is a multi-level design. The SCF (switched capacitor filter) blocks convert the modulator output to analogue.

 

DSD handling is quite different from the TI chip. In the default mode of operation, DSD input is low-pass filtered before going through the same digital volume control and sigma-delta modulator as PCM. As far as I can tell, the modulator is operated at the same rate as the DSD input, i.e. no resampling is performed. A bypass mode allows sending DSD data directly to the SCF without going through the modulator. Oddly, some versions of this block diagram show the digital low-pass filter always being active while others (the figure above) indicate that this too is skipped in the bypass mode. I don't know which is correct.

 

Finally, the ESS Sabre series found in various products ranging from the Audioquest Dragonfly to the Benchmark DAC3.

This is different from what most manufacturers do. PCM input is upsampled using a two-stage FIR filter to a maximum of 1.536 MHz. These filters are fully programmable and can also be bypassed entirely. The FIR filter is followed by volume control and an IIR filter. After this comes "THD Compensation" which I have no idea what it does. Next the data goes through an ASRC which upsamples further to a rate supposedly in the vicinity of 40 MHz (even a datasheet I'm not supposed to have doesn't say). Finally, there are the usual sigma-delta (which I assume is what hides behind the Hyperstream label) and D/A stages.

 

Like in the AKM chip, DSD input is digitally low-pass filtered and subjected to the same processing as PCM. Apparently unique to ESS is that even DSD is upsampled further, and there is no option to disable this.

 

From these examples we can see that each manufacturer has chosen a different approach. The TI design favours simplicity while ESS relies on heavy processing. AKM falls somewhere in the middle. All achieve excellent performance figures. In a final product, the surrounding electronics, notably clocking and analogue output drivers, matter far more than the DAC chip itself.

What are the advantages of further upsampling DSD?

And potential disadvantages?

Is it better to do it with hardware or in a computer by software?

1 hour ago, mansr said:

Below is a block diagram of the TI DSD1793 chip used in iFi DACs. The PCM1795 in the TEAC UD-501 is similar.

Seems like the image vanished. I'm sure it was there earlier since it's in my browser cache. Here it is again:

Perhaps @The Computer Audiophile can put it back where it belongs.

1 hour ago, jabbr said:

you've copied/reimplemented what Miska did years ago

Not accurate.  Various people have done sigma delta modulators at various times.  I believe I know the source of at least some of the basic ideas mansr used, and it wasn't Miska.

22 minutes ago, mansr said:

Seems like the image vanished. I'm sure it was there earlier since it's in my browser cache. Here it is again:

Perhaps @The Computer Audiophile can put it back where it belongs.

Done

2 hours ago, mansr said:

All achieve excellent performance figures.

Nice to know a little more about the details of how these chips may operate (in some respects depending on the specific implementation).

But as you've said, what matters are results.  Any information on comparative performance with particular DACs, software vs. internal upsampling?

12 minutes ago, Jud said:

Not accurate.  Various people have done sigma delta modulators at various times.  I believe I know the source of at least some of the basic ideas mansr used, and it wasn't Miska.

How could it be when his source code is secret? Besides, Miska didn't invent SDM.

7 minutes ago, mansr said:

How could it be when his source code is secret? Besides, Miska didn't invent SDM.

Yes, though rather than source code, I was thinking of academic articles providing some ideas you could then implement in your code.

30 minutes ago, Jud said:

Yes, though rather than source code, I was thinking of academic articles providing some ideas you could then implement in your code.

I don't recall seeing any academic articles by Miska.

2 minutes ago, mansr said:

I don't recall seeing any academic articles by Miska.

As I've now said twice: yep, not Miska.

20 minutes ago, Jud said:

 

Yes, though rather than source code, I was thinking of academic articles providing some ideas you could then implement in your code.

Right, so they way I personally apportion credit would be to the first to publish as well as the first to popularize. Some certainly for an open source implementation because people can learn from this. I've never claimed credit myself for reimplementation of algorithms -- of course I come from the days where code was assumed to be open source -- I'd certainly give credit for a visibly elegant implementation.

In any case I oppose and don't take part in arguments from authority. 

"Your wrong" without explanation is not adequate.

1 hour ago, Jud said:

 

Not accurate.  Various people have done sigma delta modulators at various times.  I believe I know the source of at least some of the basic ideas mansr used, and it wasn't Miska.

Copied the idea -- @Miska didn't invent SDM nor claimed to. His record of advocating for, and enabling and providing software for the upsampling/conversion of Redbook CD to DSD is most clearly documented on this site among other places. The record stands for itself. His implementations have given me many ideas though I have no proprietary relationship with him. I say this only to apportion credit where credit is due. 

3 hours ago, Jud said:

 

For anyone willing to do a little DIY, there's Miska's DSC1 with very good parts quality for not much over $400 and some of your time (balanced configuration under $1000).  Miska says it's the best DAC he's heard.  Let's say he's a little proud of his design and there are some very expensive DACs that beat it.  Still an interesting proposition.

 

 

Interesting, any idea if it's possible to buy a finished DSC1 for those of us without DIY skills or inclinations? 

6 minutes ago, jabbr said:

Right, so they way I personally apportion credit would be to the first to publish as well as the first to popularize. Some certainly for an open source implementation because people can learn from this. I've never claimed credit myself for reimplementation of algorithms -- of course I come from the days where code was assumed to be open source -- I'd certainly give credit for a visibly elegant implementation.

Where have I demanded credit for anything? You were the one who brought up my software along with vague accusations.

2 minutes ago, jabbr said:

Copied the idea -- @Miska didn't invent SDM nor claimed to. His record of advocating for, and enabling and providing software for the upsampling/conversion of Redbook CD to DSD is most clearly documented on this site among other places. The record stands for itself. His implementations have given me many ideas though I have no proprietary relationship with him. I say this only to apportion credit where credit is due. 

Has anyone outside the audiophile niche so much as heard of Miska? I don't mean to criticise his work, but I think you'll find his role in the greater scheme of things is rather smaller than you're imagining.

40 minutes ago, mansr said:

Has anyone outside the audiophile niche so much as heard of Miska? I don't mean to criticise his work, but I think you'll find his role in the greater scheme of things is rather smaller than you're imagining.

I have some idea what Jussi is doing. Just of curiosity, what is your product? 

1 hour ago, Norton said:

Interesting, any idea if it's possible to buy a finished DSC1 for those of us without DIY skills or inclinations? 

You get somewhat similar approach with devices like T+A DAC8 DSD and Holo Audio Spring DAC. Maybe with EMM Labs / Meitner, Playback Designs and dCS too.

22 minutes ago, AnotherSpin said:

I have some idea what Jussi is doing. Just of curiosity, what is your product? 

I don't sell a product. Why does it matter? I never claimed to be famous.

1 hour ago, mansr said:

I don't recall seeing any academic articles by Miska.

I don't work in/for academia. Nobody pays me to write such and there's nothing that would motivate me to write such, just not interested on that kind of stuff.

I'm paid to work on open source software though, and ironically that has funded most of the work done on HQPlayer which is not open source. One reason being that I find it very unlikely that I would receive much contributions from other people - thus wouldn't benefit the project...

1 hour ago, Norton said:

 

Interesting, any idea if it's possible to buy a finished DSC1 for those of us without DIY skills or inclinations? 

I have seen ads for some boards from China - don't know if they're fully populated and what the quality might be.

Since schematics and (IIRC) suggested parts lists are available,  another possibility is to pay someone local with decent electronics skills to build one for you.

13 minutes ago, Miska said:

 

I don't work in/for academia. Nobody pays me to write such and there's nothing that would motivate me to write such, just not interested on that kind of stuff.

 

I'm paid to work on open source software though, and ironically that has funded most of the work done on HQPlayer which is not open source. One reason being that I find it very unlikely that I would receive much contributions from other people - thus wouldn't benefit the project...

 

What open source software do you work on (just curious)?

6 minutes ago, Miska said:

I don't work in/for academia. Nobody pays me to write such and there's nothing that would motivate me to write such, just not interested on that kind of stuff.

 

I'm paid to work on open source software though, and ironically that has funded most of the work done on HQPlayer which is not open source. One reason being that I find it very unlikely that I would receive much contributions from other people - thus wouldn't benefit the project...

It's all good. Sorry for getting you dragged into this nonsense discussion.

5 minutes ago, Miska said:

Marantz has also used CS4398 for long time and I've used it too in my earlier designs. It supports "Direct DSD" mode (as well as top of the range Wolfson WM8741/WM8742 chips that are now part of Cirrus Logic).

https://www.cirrus.com/images/product_diagrams/cs4398blkdiag-782ad57e36.svg

 

DSD Direct mode goes straight into the SCF stage which has two options 50 kHz and 150 kHz cut-off. This gives pretty different wideband output compared the digital filter mode (I've measured both). Volume control is obviously not available in the direct mode.

Thanks for that additional information.