Which DACs bypass digital filtering?

I have been unsuccessfully searching for a list of high-performance* commercial DACs that allow user to bypass digital filtering (NOS), leaving that task to a software music player such as HQ Player.

It would be interesting if we could compile such a list in this thread.

* - no TDA1543-based DACs please, only equipment that can take 24bit and sampling rates above 176.4/192kHz.

Aqua DACs all avoid it, I think, no?

5 hours ago, semente said:

I know that the following qualify:

 

- Phasure NOS1 DAC

- Teac UD-501 and UD-503

 

How about the Lampizators?

Lampizator too.

Roch

1 hour ago, trappy said:

Aqua DACs all avoid it, I think, no?

I'll check, cheers.

Sorry for being off topic but I always assumed that if one sends a stream of max bit rate a dac supports, it automatically avoids digital filtering by a dac

For example I have a burson conductor which supports maximum bit rate of 192 khz, so if send a stream of 176 or 192 khz from my PC the filtering of dac is automatically bypassed

Someone please let me know if this right or I am completely wrong

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.

Audio-GD master 7. Can switch between nos, 2x, 4x and 8x oversampling

Re the title of this thread, I thought that oversampling and filtering were two different things?  So just because a DAC is not applying oversampling, it doesn't necessarily mean no filtering is applied?  

6 hours ago, Norton said:

Re the title of this thread, I thought that oversampling and filtering were two different things?  So just because a DAC is not applying oversampling, it doesn't necessarily mean no filtering is applied?  

Hi , even I am not sure about it and never got a clear answer for that. Maybe someone  who has gone through many dacs or is from the industry itself can clear this doubt for us

6 hours ago, Norton said:

 

 

Hi Judy

Then the question is if we feed a digital signal which is the max rate a dac can go, then will the dac still apply filtering ?

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.

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.

2 minutes ago, jabbr said:

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.

DSD256 and above allow more effective use analog filter. Because, noise pushed more far, than for DSD64/128.

I suppose any DSD DAC have no digital filter, because filtering in DSD form is impossible.

However, theoretically, there may be applied oversampling DSD for better work with analog filter.

44 minutes ago, jabbr said:

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.

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

18 minutes ago, jabbr said:

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  

You mean like ESS and AKM DACs do?

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.

6 minutes ago, jabbr said:

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.

It's still wrong in more ways than I can count.

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?

Guys 

Why fight?

It gets us nowhere