Which DACs bypass digital filtering?

[mansr]

52 minutes ago, jabbr said:

This could be difficult because you don't speak natural English as a primary language

My English is as good as anybody's.

52 minutes ago, jabbr said:

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?

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.

[Jud]

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.

[Jud]

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.

[jabbr]

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.

[Jud]

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.

[Jud]

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.)

[mansr]

1 hour ago, Jud said:

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.

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.

[mansr]

1 hour ago, Jud said:

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

They are rare for a reason. Put bluntly, they suck. Worse, they can blow up your tweeters if used with a high bandwidth amp.

[Jud]

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.

[jabbr]

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.

 

[mansr]

15 minutes ago, jabbr said:

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.

That I can agree with.

15 minutes ago, jabbr said:

the corollary being:

 

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

This, however, makes no sense. 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.

 

[Jud]

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.

[jabbr]

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". 

[mansr]

9 hours ago, Jud said:

If oversampling is applied, filtering is necessary to avoid aliasing and consequent harmonic and intermodulation distortion.

Filtering is always necessary. 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.

[mansr]

15 minutes ago, Jud said:

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.

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

[mansr]

14 minutes ago, jabbr said:

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

That choice of vocabulary was not what I objected to. Your first offence was characterising the noise as wholly digital and the signal as wholly analogue. From there it only got worse.

[Jud]

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.

[Jud]

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.

[mansr]

4 minutes ago, Jud said:

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.

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.

[mansr]

1 hour ago, Jud said:

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.

That means only that those are user-settable options. To find out what's really going on, one would need to inspect the commands sent to the DAC chip with a logic analyser. I've done this with some iFi devices, which is how I know what they're doing.

[Jud]

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?

[mansr]

20 minutes ago, Jud said:

Must an interpolation ("upsampling") filter be digital?

Yes. Although strictly speaking, sampling is a time-domain quantisation where the sample values are arbitrary, all practical storage systems use a digital representation of sample values.

20 minutes ago, Jud said:

Can a final reconstruction filter be digital?

No. A/D conversion by definition includes an analogue stage. It can be simple or complex, but there is always something.

[Jud]

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.

[mansr]

24 minutes ago, Jud said:

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.

I believe so. Why this aspect should be important is something I do not understand. What matters is the accuracy if the output. How it is achieved should be irrelevant.

[Jud]

16 minutes ago, mansr said:

I believed so. Why this aspect should be important is something I do not understand. What matters is the accuracy if the output. How it is achieved should be irrelevant.

 

I don't know the OP's reason(s).  To me, the goal of accuracy and a DAC that works in the way the OP is asking about can be related.  If we hypothesize that we have interpolation filtering and/or SDM in software producing more accurate results than can be obtained in the DAC's internal processing, then bypassing the internal processing would make sense.