Which DACs bypass digital filtering?

1 hour ago, jabbr said:

I'm pretty sure that with the correct settings, the iFi iDSD Micro bypasses input filtering.

It applies its modulator, I think (whichever of the three available you select).  But if you've already used a software modulator, I don't know how much is left to do.

3 hours ago, jabbr said:

In light of above, no DAC eschews output filtering

 

There are rare DACs that do.  I've listened to one.

2 hours ago, mansr said:

With DSD input, the filter switch selects which built-in analogue filter to use. There is no digital processing of DSD data.

Thanks, I knew it was applying a filter in that circumstance, and this makes perfect sense.

3 hours ago, mansr said:

You mean like ESS and AKM DACs do?

Are there DACs (the piece of equipment) incorporating these chips that accept DSD256/512 rates as input?

 I believe I remember ESS chips producing output at 40+ MHz rates (wasn't aware of AKM doing this), but I'd never heard of DACs with those chips accepting rates that high.  I thought ESS in particular would only allow 384KHz input max.

6 minutes ago, jabbr said:

 

Waiting for that  ... I'd say that the "output filter" is positioned farther down the chain, or else there is no concern about intermodulation distortion, perhaps for a specific reason.

The latter.  Believe me, I heard the distortion!  (My gracious host seemed either unaware or not to mind in pursuit of his goal.)

1 hour ago, Jud said:

 

Believe me, I heard the distortion!  (My gracious host seemed either unaware or not to mind in pursuit of his goal.)

16 minutes ago, mansr said:

They are rare for a reason. Put bluntly, they suck.

I believe we are ad item.

50 minutes ago, mansr said:

The AK4490EQ (among others) accepts up to 768 kHz PCM and DSD256. Its internal rate isn't as high as that of ESS, but it does process DSD inputs unless put in bypass mode. This is the chip used in the TEAC UD-503.

Looking at the UD-503 owner's manual (page 18), it appears that DSD input is handled as the micro-iDSD does.  There are two analog filter options for DSD input.

6 minutes ago, mansr said:

Filtering is always necessary.

Yes. I was answering a specific question regarding whether filtering was necessary if upsampling was done.

6 minutes ago, mansr said:

 

The point of oversampling is enabling the most critical filtering to be performed digitally where it can be far more accurate than in the analogue domain.

Can't filtering at high sample rates be analog, and digital at Redbook rates?

I thought the idea of oversampling was to allow more headroom to avoid the necessity of a "brickwall" filter.

6 minutes ago, mansr said:

Yes, but does it put the chip in DSD bypass mode?

I don't know.  However, there isn't anything in the manual to suggest DSD input rates are handled by something other than the analog filter options.

36 minutes ago, mansr said:

A sampled signal is an a representation of the original up to half the sample rate (if the original was not band-limited, the representation is inaccurate). To properly reconstruct the continuous-time signal, the images of the base band must be filtered out.

 

With a Redbook input, the images start at 22.05 kHz. In order to preserve frequencies below 20 kHz, a rather sharp (brickwall) filter is required. This is difficult to realise as an analogue circuit. Oversampling still requires the same brickwall filter, but now it can be done digitally which is much easier. A 2x oversampled signal with digital filtering has content  only up half its Nyquist frequency. With our Redbook input, the analogue reconstruction images now start at 44.1 kHz. Moreover, since only half the digital bandwidth is used, no actual images are present for another 22.05 kHz, allowing subsequent filters a full 44.1 kHz transition band. This permits a lower order, more easily implemented analogue filter. Still higher digital oversampling extends this shift from analogue to digital filtering.

 

Whenever the source is Redbook, proper reconstruction requires a brickwall filter somewhere in the chain. The choice we're given is between analogue and digital, and if accuracy is the goal, digital always wins.

Must an interpolation ("upsampling") filter be digital?  Can a final reconstruction filter be digital?

So the OP is looking for DACs that permit a PCM or DSD bitstream under one or more circumstances to be sent from input to the final analog reconstruction filter without being acted on by a digital interpolation filter.

7 minutes ago, mansr said:

What matters is the accuracy of the software plus hardware available. If the best performing solution involves some hardware processing, that should not be seen as a weakness.

Yes, best accuracy at whatever price point we're talking about.  Even inexpensive DACs seem to cost more than some fairly good upsampling and SDM software.

2 hours ago, semente said:

 

But isn't the use of analogue filters compulsory to leave out noise-shaping high frequency trash? 

An analog filter is necessary, as confirmed by @mansr.  I would say noise-shaped high frequency trash, since that's what noise shaping does, push what you don't want to higher frequencies.

2 hours ago, semente said:

 

Could you elaborate a bit on the hardware processing for PCM and DSD? 

I too would like to see information on superior hardware processing (DAC chips, FPGAs, shift registers, whatever) at prices competitive with available software, if @mansr knows of anything that has been implemented in consumer, pro, or DIY equipment.

34 minutes ago, Norton said:

 

I'm listening now to HQP via NAA into my Esoteric DAC with (as per the TEACs) its onboard upsampling and filtering options set to off.  Sounds good, but is bettered to my ears just fed by my BDP, letting the Esoteric do the up sampling and filtering (although admittedly not when using the DSD conversion option)

 

In terms of superior hardware processing, I borrowed  a DAVE a few weekends back.  The results were superb, better than any digital  I've heard to date.   However it's an expensive  item and I'm interested in how close I could get instead with a DAC specifically tailored to HQP.

 

I can't see though that comparing software vs. DAC prices is meaningful.  The software is just upsampling and filtering, surely the analogue conversion, power supply and output  stage of a DAC make up 50% plus of eventual SQ?

 

 

 

I tend to think of a DAC as one third digital (filter) design, one third analog circuit design, and one third parts quality.  So if one can get a DAC with reasonable analog circuit design and parts quality, and substitute software for the cost of digital filter design, cost efficiencies for you as a consumer might be realized.

I'm guessing the DAVE's filtering, implemented in an FPGA, might be pretty well simulated by someone who knew what he or she was doing with the excellent iZotope software and @mansr's SDM, available in Audirvana+ for I think $79 or so.

Just now, rickca said:

I agree you can simulate a filter in software, but those 3 elements aren't really independent.  There's an overall architecture/design that brings it all together that determines the ultimate performance of the DAC.

I don't disagree.  But I've got a $375 DAC that uses commodity DAC chips, nothing special, and I use software upsampling with it to get better sound at far less expense than would otherwise be possible for me.  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.

I'm never averse to trying to gain and use a little knowledge to get something better for less expense.

2 minutes ago, jabbr said:

 

Many equations are known and algorithms available and have been implemented in hardware for a considerable time. FPGA is a variant of hardware. Not sure who first implemented upsampling in software but at least for our purposes I credit @Miska for developing the HQPlayer software implementation for DSD (it does PCM also) and @PeterSt for the XXHighEnd for PCM, and the so named NOS1 as its intended DAC. I think price point contains many factors but given what they do the software packages are terrific bargains.

The software packages also provide more flexibility than commonly found in hardware.  (Of course this allows you to screw up, too.) The iZotope SRC in A+ is used in recording studios (costing $300-$700) and offers adjustable parameters.  A+ also provides a choice of mansr's modulators.  Miska offers quite a variety of filters and modulators.  And Peter does offer custom filters as well as his own single one (I've never been motivated to try a custom filter with XXHE - I like Peter's).

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.

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?

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.

2 minutes ago, mansr said:

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

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