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.

 

 

 

 

 

 

 

 

 

 

[Miska]

8 minutes ago, numlog said:

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?

 

From the DAC point of view, not so much, for example the Sabre 9038 has 64 unity-weighted elements. DSC1 has 32. If you look at other discrete implementations like dCS RingDACs, they have 24. Chord varies from 4 to something like 20. The rest is about how the element array is used and what kind of analog reconstruction filter there is at the output.

 

Large portion of the performance is defined by the DSP algorithms generating the bitstream. The DAC itself is fairly simple. That is one of the purposes of SDM DACs, that as much as possible things that require accuracy and complexity are done in digital domain using DSP, rather than in analog domain on hardware. This improves reliability and precision while also being cheaper at the same time.

 

11 minutes ago, numlog said:

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?

 

Complexity of the algorithms. Things like to which rate the digital filters are run to and how the modulators operate. These algorithms can be quite different.

 

13 minutes ago, numlog said:

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.

 

With those chip-converters, you again have similar limitations as you have with the built-in converters of DAC chips like 9038 and the exercise doesn't have much point.

 

Modern DAC chips are mostly designed to a price point, so they are generally below 20EUR/piece. And when necessary, cut some corners to achieve that.

 

[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

[Miska]

1 hour ago, numlog said:

only things start to get confusing when using HQPlayer's best filters werent able to outperform the DACs internal DSP.

 

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.

 

1 hour ago, numlog said:

I have wondered how the 9038 DACs handles the DSD input before the SDM portion, for example it does allow volume control.

 

It goes through it's DSP with volume control and remodulation. So it doesn't help working around it's modulator. But it helps working around shortcomings of it's digital filters. Or put other way, DSD goes through much simpler DSP chain in the chip than PCM does.

 

1 hour ago, numlog said:

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

 

Yes, down side of Sabre is that it doesn't have Direct DSD mode. While AKM, Cirrus Logic and Wolfson have. And TI/BB chips don't even have anything else.

 

You can find some DSC1-style DACs on the market that have a separate discrete DSD section. First and most well known is T+A with DAC8 DSD and the newer more expensive models. I guess Denafrips is closest to DSC1. Then there are Holo Audio DACs which I believe are also similar and (AFAIK) also feature AKM SRC chip you mentioned (just like TEAC UD-50x series).

 

Thread here discusses some of these aspects (but also about R2R PCM implementations):

At the end you can find fairly up to date list of DACs.

 

[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).

 

 

 

 

 

[PAR]

I have been able to transcode PCM to DSD for most of this century using dCS upsamplers and DACs. Initially impressed I have grown to dislike it so much that I now have a dCS upsampler packed away in its box in my spare room. BTW, dCS introduced new algorithms for this conversion with the Vivaldi range. Yet I still don't like it compared to unvarnished PCM when an appropriate digital filter has been selected as dCS equipment allows.

[Miska]

10 minutes ago, PAR said:

Yet I still don't like it compared to unvarnished PCM when an appropriate digital filter has been selected as dCS equipment allows.

 

dCS is never "unvarnished PCM" because it is always an SDM DAC by construction. So it goes through modulators always, just different modulators. It technically cannot play PCM without conversion.

 

With DSD upsampling the choice of digital filter is as important as for any other case too.

 

One limitation of dCS' DSD-upsampling is that it produces usually DSD64 (2.8 MHz) or at most DSD128 (5.6 MHz). These are fairly low rates these days. While I can play up to DSD1024 (49.2 MHz)...

[PAR]

55 minutes ago, Miska said:

One limitation of dCS' DSD-upsampling is that it produces usually DSD64 (2.8 MHz) or at most DSD128 (5.6 MHz).

 Absolutely correct.  But in my view, so what? Do ever higher rates make the music any better ? They may improve the sound marginally but that isn't my question. 

 

I know that is an unfair question for you as you have what Churchill called " a dog in the fight" given your business profile. You have to say " Yes" of course 😉.

 

 

 

 

 

 

 

 

[jamesg11]

An odd question, though, on an ‘audiophile’ site?

[Miska]

6 hours ago, PAR said:

 Absolutely correct.  But in my view, so what? Do ever higher rates make the music any better ? They may improve the sound marginally but that isn't my question. 

 

I know that is an unfair question for you as you have what Churchill called " a dog in the fight" given your business profile. You have to say " Yes" of course 😉.

 

Yes, in my opinion step from DSD64 -> DSD128 is huge, from DSD128->DSD256 still big, and then it starts getting smaller.

 

Have you tried yourself DSD64 -> DSD128 -> DSD256 -> DSD512? If you haven't, you cannot say "No" to your own question either.

 

And not the rates alone, also the modulator choice matters. So it is good to listen to at least couple of modulator alternatives.

 

[PAR]

3 hours ago, jamesg11 said:

An odd question, though, on an ‘audiophile’ site?

Not if you ask what the purpose of audiophilia is - or should be.

[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.

[jamesg11]

8 hours ago, PAR said:

Not if you ask what the purpose of audiophilia is - or should be.

I alluded to the music-sound antinomy at the locale of the ?-mark

ie Miska’s follow-up didn’t answer such a ‘question’ either, or could hope to.

[Miska]

On 11/21/2019 at 4:44 PM, numlog said:

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

 

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.

 

On 11/21/2019 at 4:44 PM, numlog said:

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.

 

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.

 

[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?

 

 

[Miska]

10 hours ago, numlog said:

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?

 

More elements mean earlier roll off and more attenuation. You double the number of elements and the roll off starts at half the frequency. The roll off point is relative to sampling rate, so the absolute frequency depends on the sampling rate.

 

[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.

[luisma]

On 11/19/2019 at 5:03 PM, Miska said:

Or put other way, DSD goes through much simpler DSP chain in the chip than PCM does

I would have thought this DS DAC would perform / output better sound when fed by DSD vs PCM, your comment of much simpler DSP chain means "better" or "worse" with DSD? I understand it as better but @numlog

 

On 11/19/2019 at 11:21 PM, numlog said:

there is probably not much better than the 9038Q2M for PCM

So this last comment got me confused

 

@Miska by the way on your Project S2 which filter do you use? fast roll off ?

[Miska]

48 minutes ago, luisma said:

I would have thought this DS DAC would perform / output better sound when fed by DSD vs PCM, your comment of much simpler DSP chain means "better" or "worse" with DSD? I understand it as better but @numlog

 

 

Yes, better... But not sure about the context here...

 

48 minutes ago, luisma said:

@Miska by the way on your Project S2 which filter do you use? fast roll off ?

 

If you send there 705.6/768k PCM, the filter selection has no effect since the built-in filters get bypassed.

 

[luisma]

6 minutes ago, Miska said:

 

Yes, better... But not sure about the context here...

 

 

If you send there 705.6/768k PCM, the filter selection has no effect since the built-in filters get bypassed.

 

Summarizing, if using HQPe send DSD to the DAC, not PCM.

 

Are the DAC's FIR filters always bypassed when sending DSD?

 

[Miska]

21 minutes ago, luisma said:

Summarizing, if using HQPe send DSD to the DAC, not PCM.

 

Are the DAC's FIR filters always bypassed when sending DSD?

 

Do you have a specific DAC in mind?

 

[luisma]

I'm sorry it wasn't clear before, just talking about the Project S2 DAC which you mentioned above

[Miska]

Just now, luisma said:

I'm sorry it wasn't clear before, just talking about the Project S2 DAC which you mentioned above

 

Yes, in that case there is no digital oversampling FIR in play when you input DSD.

 

Maybe the confusion is about digital oversampling FIR vs analog D/A conversion FIR? These two are completely different things. For example my DSC1 design discussed here has analog FIR D/A conversion stage, which is fundamental thing for it's operation.

 

[luisma]

15 minutes ago, Miska said:

Maybe the confusion is about digital oversampling FIR vs analog D/A conversion FIR?

Thank you for clarifying, I was just referring to digital oversampling filters, I do have questions about the D/A stage conversion FIR but that's for another thread / time, don't want to be OT here

 

[Andy-HandyHifi]

On 11/25/2019 at 1:22 PM, 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.

 

 

Hello Miska

 

This is the first time I have been on this site in many years.  At the time using another name.  At that time I was skeptical of your claims and software.  All these years later I have studied DSD, Multibit Delta Sigma, Thermometer/Unary Code, FIR filter design on both the digital and analog level, delay lines, moving average filtering, taps, even down to how 'bits' mean different things in different contexts and can get people totally confused.  

 

Funny thing after all these years of study, I came back and re-read your posts on your open source DAC and your software, and, how about that... all the conclusions I had made during those years were right there in front of me the whole time.  

 

I suppose we all have to go on our own journey of 'understanding'.  I am certainly still on my journey of understanding.  The more we learn, the less we really know, so they say.  It is really a truth.  

 

I would only clarify a couple things I read in the last few threads.  I believe you intended to say that the BB/TI chips while not offering a DSD- remodulator with volume control ala AKM, Wolfson, Cirrus, etc, it certainly offers a 'DSD Direct' mode.  It has a 'DSD ONLY' mode ;) when not converting PCM.  That being, when a DSD signal is present, the ONLY way it is processed is the direct method.  I am positive this is exactly what you were saying.  I thought that may have been lost on some people.  

 

The TI/BB chips like iFi still uses (for various reasons that they were once too happy to explain but now they have grown so much, not near as much time to explain :) ) essentially the same method you do as well as several others.... 

 

Burr Brown/ TI : 1 bit DSD bitstream filters the  bitstream via an 8 tap/8 level/8 bit FIR filter (see the confusion about bits and taps, levels, etc?  We are not in two's complement PCM here folks.. rather in unary code which can be very well be envisioned as multiple 1 bit streams now in parallel. Those 8 bits of the DSD bitstream are time delayed by a clock cycle til you get 8 identical bitstreams exactly 8 'bits' long, making a rather ingenious 8 bit/9 level moving average FIR filter whose taps represent the desired voltage/current value all summed together at the output to produce to desired final analog current/voltage.  Do this billions of times, and VOILA!  You have your analog DSD conversion to enjoy hearts content! (of course in the end each bitstream is more than 8 bits long each, that process continues ad infinitum till finished.  I was just trying to paint an easily understandable image)

 

 

Not to speak for Miska, but I believe his open source design acts in a similar way but with a 32 bit/tap/level FIR filter.  Still no 2's complement PCM to be found here.  To elaborate further,  actually since we are in Unary code where we cannot have both a positive and a negative 0, and actually have no negative numbers here at all... All taps/switches are turned off giving a value of 0.  So we had taps/switched numbered 0-32 for 33 levels or bits!  If we thought of it as we commonly do in 2's complement PCM with negative values, we get this... (assuming we have negatives in unary code... we dont)

 

16,15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0,-1,-2,-3,-4,-5,-6,-7,-8,-9,-10,-11,-12,-13,-14,-15,-16 

 

but 2's complement PCM has a 'sign' bit that tells us negative versus positive, and that 'bit' is the MSB, so PCM of the above would look like this..

 

15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0,-1,-2,-3,-4,-5,-6,-7,-8,-9,-10,-11,-12,-13,-14,-15,-16 

 

giving you the the 32 values one would expect of  two's complement 5 bit PCM.  (which of course becomes MUCH MUCH more efficient a way to represent the values once we give to millions of levels in 24 bit and greater ;) 

 

Of course my knowledge in this area of PCM theory, sign bits what to do with the extra value of -16 when stripped of the +16 counterpart for sign duty, is very, very poor.  Once gain, the more you learn the less you know.  

 

 

 

My only minor disagreement in what I have independently discovered is that the Signalyst DAC is a 32 level DSM.  When it comes to DSD, it is just a 32 tap FIR filter.  Yes it is the same 1-bit DSD bitstream presented simultaneously offset by a clock sample 32 times to accomplish this filter.  

 

My agreement is why the heck don't we do this all the time??? Why can't be have SDM modulators (other than the Grimm 1 bit converter, discounted here for obvious reasons) output their 4, 8, 16, 32, 64 or 128 level modulator output as a standardised file to be edited, mixed, etc, and THEN downsampled or decimated to the appropriate final format?  Are there ANY ADC's out there that directly output their 'multi-bit intermediary' format directly out for the ultimate in audio format quality?  

 

 

Again, I come with apologies for thinking I knew way too much at the time.  I gues we all have a bit of 'know it all' at a time or two.  And like I said, the more I learn, the less I know.  

 

Thank you Miska.  Although the FPGA in my iFi iDSD Pro sounds exceptional, the Signalyst simply by is nature is more capable and I will be using it from now on with Roon/HQPlayer and iDSD direct DSD decode at DSD 1024.   

 

Andrew