PCM to DSD (DSC1) vs native PCM

[numlog]

For Sabre 9038 DACs I found the best SQ came from native PCM playback. No upsampling or DSD conversion, there was clearly some benefits to these options but negatives too, native PCM seemed the most balanced in the end.

 

The DSC1 DAC is designed to deliver a DSD input to the output with as little processing as possible, since not much is needed for DSD.

For native DSD there is probably not a better choice of DAC than DSC1... but what about PCM?

It appears the DSC1 is also recommended for use with PCM to DSD conversions.

My understanding is fairly limited but AFAIK converting PCM to DSD is similar to how a Delta Sigma DAC works.

How do they differ? Why would one be superior to the other?

Based on listening experiences, did any of you find one of them to be superior?

 

At a given DSD rate, DSD to PCM conversion can be done very with little computing power or it can require more computing power than what our fasted consumer CPUs are capable of.

what is the difference with these conversions?

 

If you aren't using a PC the option for PCM to DSD conversion would be limited to the AK4137 SRC.

I am interested in the DSC1 for DSD playback regardless but I could pick up an AK4137 board for PCM conversion...

The question is if there would be any point if I already own a decent 9038Q2M DAC when the aim is to achieve the best PCM playback possible.

 

 

 

 

 

 

 

 

 

 

[numlog]

So there really is not much difference between the DSC1 and SDM portion of 9038 DAC. This was the theory behind why you should use HQPlayer, only things start to get confusing when using HQPlayer's best filters werent able to outperform the DACs internal DSP.
I have wondered how the 9038 DACs handles the DSD input before the SDM portion, for example it does allow volume control.

AKM DAC's have some sort of bypass mode for DSD that bypass volume control, filtering and whatever else before SDM stage, clearly there must be some disadvantage to what Sabre is doing for DSD if AKM has this mode, so the DSC1 might be the only right choice to hear HQPlayer to its full potential

[numlog]

 

6 hours ago, Miska said:

Which way?

 

On my 9038Q2M based DAC (Pro-Ject PreBox S2 Digital), I have confirmed results of improved digital filter performance and some other aspects. You can find the results from the relevant thread here.

Yes I seen them, infact they led me to explore DSD conversions in the first place, I only meant subjective performance but probably better not to go down that road.

 

So anyway without a PC to perform conversion there is probably not much better than the 9038Q2M for PCM.

R2R is also interesting, the concept certainly appeals to the audiophile, being the ''purest'' way to handle PCM, but does not carry any theoretical advantage.

There have been audiophile beliefs that were complete nonsense/exaggerations (e.g negative feedback and op amps should be avoided), which even seem somewhat rational, and others that held true but make no sense at all (which has been particularly common in computer audio).

 

 

 

 

 

[numlog]

What is the resulting difference (subjective and objective) between having 32 elements and 64?

The reason I ask because the DSC1 boards you can get from china are the updated version with differential output and 32 elements per phase.

It uses XOR gate to get the inverted DSD signals and then a transformer to convert the hot and cold output to single ended...

I am not sure the reasoning behind this but doubling the component count plus adding an XOR gate and especially a transformer to the signal path just to end up with the same single ended output as the original design seems counter intuitive.

So how about combining all 64 elements per channel and leaving out the XOR gate and transformers? this would closer resemble the 9038 SDM conversion

 

Alternatively what about having far less, 4 or 8 elements? I am weary of quality of chinese kit so also considering building a simpler version from scratch... it would cost basically nothing so no reason not to.

[numlog]

3 hours ago, Miska said:

 

Mainly roll-off of the resulting analog FIR filter. My 32-element DSC1 design works OK starting from DSD128 (not too much roll-off at 20 kHz yet). If you intend to run higher rates, then you can increase number of elements as well.

 

 

I would say that is not enough with unity weighting, but you could cut it down to 16- or 24-elements if you like, especially if you plan to run it at DSD64. But you really should look it as entire DAC including the analog reconstruction filter following it, which in my case is 4th order. IOW, what matters is combined performance of the analog FIR DAC + analog reconstruction filter.

 

Also the D/A conversion stage design affects jitter sensitivity.

 

Unless im interpreting it wrong it sounds like more elements = slower roll off?

I was assuming it would create a steeper filter, but also suspecting it would influence THD and noise performance, so if it doesnt improve THD/noise or make the filter more effective what's the purpose?

 

 

[numlog]

Right, that makes things clearer.

The thread on diyaudio touches on some of these things so far, I will read through and hopefully learn a bit more.

 

It will need to be adjusted for DSD64 i.e native SACD rips.