Which DACs bypass digital filtering?

[jabbr]

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

[jabbr]

This is my understanding:

(I'm going to limit the discussion to DSD in this post and can discuss PCM separately if desired -- just ask)

 

The "sound" is contained in the digital recording. The goal of the reproduction system (DAC + Amp + Speakers) is to accurately product the "sound".  During the playback process, the "sound" is mixed with "noise". In a DSD (SDM) bitstream, the "sound" is directly contained in the "analogue" part of the bitstream, the "noise" is contained in the "digital" clock that is used to transport the stream from one place to another. The function of the DAC is to separate the analogue sound from the digital noise.

 

This is really really simple, so if you don't understand what I've written above, go back and reread, because understanding this is essential to understanding the process. The last sentence, in particular, accurately and specifically describes the function of the DAC.

 

In DSD/SDM the digital noise is contained in the carrier clock (BCLK) as well as its harmonics. The BCLK is necessary to interface the analogue signal with the digital system and the goal of the DAC is to remove all vestiges of the BCLK from the analogue signal without disturbing the signal itself. This where upsampling and filters come into play.

 

Let's say we allow everything to pass including the carrier BCLK -- we can't hear it right? Speakers can't reproduce it right? What's the big deal? That's where intermodulation distortion comes in: high frequency noise interacts with the electronics to produce measurable, audible and very harsh sounding distortion in the audible band.

 

One might consider a "brickwall" filter which would allow the analogue signal to pass and cut off everything above what we define as either 44 kHz or 96 kHz or whatever we define as the upper limit of the analogue signal we want.

 

Well it turns out that these "brickwall" filters also have distortion that extends below the cutoff frequency: the brickwall filters aren't perfect. So a much much better idea is to use a gentle filter at the corner frequency but in order to get the gentle filter to effectively filter out the digital noise we need to "noise shape" which is where the upsampling comes into place: the upsampling increases the frequency of the digital carrier clock (BCLK) thus increasing the frequency separation between the analogue signal and the digital noise and thus improving the ability of the gentle filter to remove the noise. Viola'

 

Now 99% of PCM starts out as SDM/DSD and ends up as SDM/DSD to the same argument applies with the added complexity of where, when and how to convert between SDM and PCM.

 

 

[jabbr]

In light of above, no DAC eschews output filtering

 

Typically if a DAC accepts DSD256,512 its going to be very difficult to realtime filter that signal! so I'd assume there's no input filtering, but if there is, someones done a great engineering job  

 

Any DAC that does input upsampling does input filtering.

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30 minutes ago, mansr said:

Sorry, but that makes no sense whatsoever. I suggest you study the maths involved properly before trying to explain things.

 This is intended to be a simple natural English language explanation for people who speak English not a mathematical explanation for people who speak math. Naturally the English language is subject to human interpretation but you might go back and learn how to communicate with humans.

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1 hour ago, jabbr said:

The function of the DAC is to separate the analogue sound from the digital noise.

 

This could be difficult because you don't speak natural English as a primary language and I understand that you are having trouble counting the ways it is wrong so let's start with a simple sentence. This is an English language sentence. Do you understand? Do you agree? Do you need it spelled out?

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2 minutes ago, Jud said:

 

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

 

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.

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2 hours ago, mansr said:

My English is as good as anybody's.

I understand the words perfectly. They simply do not have a meaning which can be considered correct in any scientific or engineering sense. If I were to explain a DAC in simple terms, I'd go with something like this: The function of a DAC is to convert a digital representation of sound to an analogue form.

 

That's a dictionary definition.

 

I said:

Quote

The function of a (DSD) DAC is to separate the analogue signal from the digital noise

 

In the simplest engineering terms I can think of, and assuming a single DSD channel switching between 0 and 5v:

 

The simplest (DSD) DAC is nothing more than a low pass filter.

 

the corollary being:

 

The function of the low pass filter is to remove the high frequency digital noise from the analogue signal.

 

One can spend endless time debating the details of the LPF, however the two specific implementations I will point to (as if this needs to be physically proven but here we are): @Miska's Signalyst DSC1 and the "No DAC". I've heard that the Lampizator uses the same approach using tubes for the filter, but don't have an actual schematic.

 

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3 minutes ago, mansr said:

...As long as the DSD stream is seen as a sequence of bits, both signal and noise are digital. When seen as a varying voltage, signal and noise are both analogue. Since the bulk of the noise has been separated in frequency from the signal of interest, it is possible for a low-pass filter to remove the former while retaining the latter.

 

 

Ah, ok. I am using the (admittedly nontechnical) term "separate" to mean "remove" as in "remove the noise while retaining the signal". 

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3 hours ago, semente said:

Thanks.

I would also be interested in reading the PCM discussion if you feel like writing about it.

 

PCM is conceptually different from SDM (DSD) in that the analogue signal itself has been converted to a digital signal (numbers). DSD stream does not directly contain numbers -- the analogue signal remains present being at the low frequencies while the digital carrier is present at the higher frequencies.

 

PCM is a stream of numbers -- in this case the DAC has to take the numbers and convert to an analogue voltage, rather than (simply) filtering out the high frequency carrier/noise. Classically an R2R resistor ladder is used, and the analogue output filter removes the remaining aliasing noise.

 

I am not going to focus on the implementation details of A/D or D/A conversion as implemented by various chips or software, rather the benefit of upsampling. I'd say that upsampling is even more important for PCM than DSD for the following reason:

 

In PCM, the aliasing noise has its largest component at the sampling frequency and this is what needs to be filtered out. Thus the brickwall filter. The "problem" with analogue filters is that they typically have effects not only on the frequencies above the corner frequency but also at frequencies lower than the corner and so with a corner frequency of 44.1 kHz, the output filter clearly has effects on the analogue signal itself. Not good. By upsampling PCM, this is the same as "noise shaping" DSD in that the aliasing frequency is pushed higher and similarly the analogue output filter may then have less effect on the desired signal. 

 

Of course if the DAC first converts the PCM signal to SDM, then the issues described for DSD apply.

 

I am using the term "digital noise" not in any sort of perjoritave sense that "digital is bad" rather to express from the point of view of the amplifier, speakers etc, this these frequencies, which are artifacts of the digitization process, are what need to be removed by the output filter. I could simply call this noise, but the term "digital noise" is intended to reflect the fact that the principal component is at the digital clock frequency. By upsampling whether PCM or DSD, this clock frequency is increased without changing the signal frequencies thus reducing the effects of the required output filter on the analogue signal itself.

 

Hope this is helpful, and I hope this post hasn't caused anyone to choke on their cheerios  

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6 minutes ago, Jud said:

 

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.

 

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.

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3 minutes ago, mansr said:

 

 In a final product, the surrounding electronics, notably clocking and analogue output drivers, matter far more than the DAC chip itself.

+1 this is true because implementations up to the analogue point are uniformly high. 

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5 minutes ago, mansr said:

This is a common misconception. I really ought to do a proper write-up explaining how these things actually work.

Not a misconception at all. Are you able to defend yourself in English?

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6 minutes ago, Jud said:

 

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).

I don't frankly fool around with filter parameters, at least yet. I'm sure there are software libraries to come with new filters for new applications. I have no doubt that HQPlayer could be copied with enough time and energy, but the ability to load room correction and other kernels being processed in the SDM domain is terrific. You could develop these kernels in external packages like acourate (iirc).

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1 minute ago, mansr said:

Which part of my English do you have trouble understanding? Do I need to dumb down my vocabulary for you?

Specifically what is the "misconception" and don't use equations or drift off into irrelevant hardware diagrams. Are you able to write plain English --- if find no evidence that you have that capability. 

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37 minutes ago, mansr said:

 

Aliasing is not noise and has nothing to do with noise shaping. Also, your filter frequencies are off by a factor 2.

Try not to argue by authority -- I don't consider that you've copied/reimplemented what Miska did years ago anything that grants you authority in my book.

 

"Aliasing is not noise" -- I never stated that. I have defined the term "noise" explicitly. Never said that aliasing has anything to do with noise-shaping.

 

Perhaps you are having a problem reading & understanding? 

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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.

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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. 

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25 minutes ago, semente said:

 

Is it just a hardware thing or does it require programming? 

It is literally a USB interface (Amanero) followed by a digital FIR filter (chip registers) followed by an analogue low pass filter. That's it. Simple.Dsc1.pdf

[jabbr]

Edit above "chip registers" -> "the chips are shift registers"

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17 minutes ago, Jud said:

 

With the proviso that hardware in some intermediate state may be available from China, yes, that's correct.

 

Uh, since the Chinese source:

1) has a different circuit, which is allowed under @Miska CERN Open Hardware License

2) however doesn't publish the circuit, which is entirely against the CERN Open Hardware License, I'm not going to discuss nor promote the effort.

3) Who knows what it is? Says DSC1 but who knows...

 

17 minutes ago, Jud said:

 

 

A group buy of boards populated with the parts that would be more difficult to DIY has been discussed.  This would make it easier and cheaper to have some local person finish it.  And a group buy could include some or all other parts and a chassis, making up a kit.

 

But nothing's gone past the preliminary discussion stage.

 

It would be possible to group buy a set of finished circuit boards but there would need to be a number of people interested to make this cost effective. This would be the parts as described. The production facility can then populate the boards. This is particularly good for the SMD parts but can also be done for TTH. The design includes a power supply, and uses the Amanero USB interface all of which could be grouped together, and a case etc.

 

That said, I've already built the DSC1 and am personally more interested in modifications. Since I have a good day job  this is a hobby for me, I enjoy designing and laying out circuits, and and have become very interested in using the Zynq as a universal interface for pre-amp/DAC whatever, particularly to build an Ethernet interface... programming an FPGA requires certain skills (I used this project to learn) and currently slogging through writing a device driver (another skill that I am trying to learn) ... this would all be to replace the Amanero with something better   I think the fundamental DSC1 design is terrific and readily modifiable and await @Miska's promised DSC2 design... he also has many other things on his plate... so in any case if someone wanted to organize a group buy I could assist, but that's not what I do best as a primary skill.

 

That said, I think the very best thing about the DSC1 is that it is a fantastic way to learn exactly what a DSD DAC needs to be -- without frills, and if folks are interested we could start discussing the DSD DAC from a technical/mathematical aspect using the DSC1 as a concrete and easily understandable archetypical example. I could start the discussion in another post so as not to put this very interesting thread completely off topic.

[jabbr]

On 5/29/2017 at 10:54 AM, mansr said:

Got any more insults while you're at it?

My sincere apology for being insulting. I was attempting to avoid an argument by authority and even if I felt insulted myself that is not an excuse for going on the attack -- I did not mean to denigrate your contributions, and I particularly respect your open source contributions. I do hope that these will continue to lead to commercial opportunities. My apologies.

[jabbr]

59 minutes ago, Jud said:

 

Apparently what Trinity says is that "8x oversampling" is achieved by having 8 PCM1704 chips per channel and having the appropriate phase delays in 7 of them to get 8 output samples per input sample.

??

[jabbr]

It is, of course, a digital filter, 8 bit FIR I guess (depending on how its connected), but upsampling? no.