Null test 88.2/24 and 44.1/16

[Jud]

Just now, pkane2001 said:

 

The ESS DAC, yes, but not the R2R one

 

 

 

Right, but you never played DSD on that one anyway.  

[mansr]

7 minutes ago, Jud said:

Although maybe you kind of haven't, since your DACs very likely do sigma-delta modulation internally.   

Even if his DAC doesn't, the ADC almost certainly did.

[pkane2001]

25 minutes ago, Jud said:

 

Right, but you never played DSD on that one anyway.  

 

Why? Been playing DSD on both DACs for at least 6 months straight (both, PCM and DSD content) until I tried switching back to PCM. The R2R DAC I have is using a resistor ladder for both, PCM and DSD playback. Mansr's point that nearly all PCM is captured using sigma-delta ADC process is valid, though

 

 

[pkane2001]

2 hours ago, pkane2001 said:

As to the comparison with 24/96 PCM recording,  I take DSD256 and then convert it to 24/96 for analysis. Possibly whatever benefit there exists in DSD due to less aggressive filtering and potentially higher resolution is lost in this process? Not sure. I’ll try to convert to higher sampling rates to see if it makes a difference.

 

So I upped the DSD256 conversion rate to 192KHz for comparison. The PCM track was also resampled to the same rate since the original was at 96Khz. Here's the result:

 

Here's the original 96KHz sample-rate comparison, for reference:

 

[mansr]

I just noticed something. Apparently Sound Liaison have abandoned the DAC/ADC method for DSD conversions, or at least on this album they used the Audiventory software converter. This program, bizarrely, applies a low-pass filter to everything it processes even if not necessary for the operation. That explains the oddities seen here.

[Jud]

1 hour ago, pkane2001 said:

 

Why? Been playing DSD on both DACs for at least 6 months straight (both, PCM and DSD content) until I tried switching back to PCM. The R2R DAC I have is using a resistor ladder for both, PCM and DSD playback. Mansr's point that nearly all PCM is captured using sigma-delta ADC process is valid, though

 

Ahh, only ever recall reading about resistor ladders in DACs using PCM as input. Learn something new...

[tmtomh]

27 minutes ago, mansr said:

I just noticed something. Apparently Sound Liaison have abandoned the DAC/ADC method for DSD conversions, or at least on this album they used the Audiventory software converter. This program, bizarrely, applies a low-pass filter to everything it processes even if not necessary for the operation. That explains the oddities seen here.

 

Wow, that's rich. So the guy who's obsessively banging on about using the "best" optical-disc drives and software to get accurate rips of CDs without using the AccurateRip database, has written DSD conversion software that unnecessarily adulterates the musical data?

[mansr]

1 hour ago, pkane2001 said:

The R2R DAC I have is using a resistor ladder for both, PCM and DSD playback.

That doesn't make sense. Which DAC are we talking about?

[pkane2001]

1 minute ago, mansr said:

That doesn't make sense. Which DAC are we talking about?

Holo Spring.

[mansr]

8 minutes ago, pkane2001 said:

Holo Spring.

I found this "explanation" on the internet:

"The DSD module of Spring does not use R2R architecture for the DSD, more specifically it uses a very specific architecture which is optimized to perform DSD to analog."

 

My guess is that it uses a parallel-out shift register with analogue summing of the (possibly weighted) taps, much like Miska's design.

[Jud]

31 minutes ago, mansr said:

That doesn't make sense. Which DAC are we talking about?

 

While this is in line with what I've read, I'm interested in knowing why it doesn't make sense.  Why doesn't a resistor ladder work with DSD (or isn't that the thing that doesn't make sense?)?

[mansr]

7 minutes ago, Jud said:

While this is in line with what I've read, I'm interested in knowing why it doesn't make sense.  Why doesn't a resistor ladder work with DSD (or isn't that the thing that doesn't make sense?)?

In the simplest design, there is only one switching element. If that is a ladder in the electrical sense, then so is a simple wooden stick in the traditional. A moving window with analogue summing could be implemented using a ladder structure, but it wouldn't be R-2R. If a DAC billed as R-2R handles DSD, it is either converted to PCM, or there is a separate circuit for it.

[pkane2001]

7 hours ago, pkane2001 said:

 

So I upped the DSD256 conversion rate to 192KHz for comparison. The PCM track was also resampled to the same rate since the original was at 96Khz. Here's the result:

 

 

Here's the original 96KHz sample-rate comparison, for reference:

 

 

 

And if anyone is still watching... Here's the Blue Coast Music track, DSD256 converted to 24/192 (blue) compared to 24/96 converted to 24/192. The falloff at 48KHz in the PCM track is due to the Nyquist frequency  filter applied as part of the upsampling process:

 

 

[mansr]

3 minutes ago, pkane2001 said:

And if anyone is still watching

Could you please make the frequency axis tick marks a little less random?

[pkane2001]

Just now, mansr said:

Could you please make the frequency axis tick marks a little less random?

 

Not easily. The charting package is selecting these for me, something I've struggled with for a while (oxyplot). I can zoom in on any area of the chart, and it'll fill them in a bit better

 

 

[mansr]

5 minutes ago, pkane2001 said:

Not easily. The charting package is selecting these for me, something I've struggled with for a while (oxyplot). I can zoom in on any area of the chart, and it'll fill them in a bit better

What the hell were they smoking?

[pkane2001]

25 minutes ago, mansr said:

What the hell were they smoking?

 

good question! Looking at the source code, they spent a lot of time on this logic to make it as automatic as possible...

 

[mansr]

43 minutes ago, pkane2001 said:

good question! Looking at the source code, they spent a lot of time on this logic to make it as automatic as possible...

Someone should show them the rand() function.

[fas42]

2 hours ago, mansr said:

What the hell were they smoking?

 

The developer of LTspice seems to be on the same stuff ... if you pan an output plot you get the same craziness, meaning you have to zoom in and out to restore sanity to the axes. I've mentioned this several times in bug reports, but still no joy ...

[bachish]

17 hours ago, pkane2001 said:

 

I think that’s true, bachish, but I’m not sure I proved the general care here, just a couple of recordings.

 

The first was obviously run through a sharp low pass filter around 24KHz, so there’s almost no useful information above that point. Whether information above 24KHz is ever useful is also certainly worth considering

 

DSD quantization noise is huge in the audible range, so it needs to be shifted to higher frequencies through noise shaping. Once there, I think it’s still a very good idea to filter it, otherwise there will be a lot ultrasonic noise that can interact with the components in your system, and some say, even intermodulate into the audible range. So, if filtering is necessary, the next question is where to do it.

 

24 kHz seems reasonable enough, although I don’t see the point of using such a sharp filter as in the recording from Sound Liaison. As Mansr said, perhaps they are using some equipment that has this filter without realizing it’s there. I’d move it further out and make it a more gradual filter. At least that might allow some possible benefits of DSD to come through.

 

As to the comparison with 24/96 PCM recording,  I take DSD256 and then convert it to 24/96 for analysis. Possibly whatever benefit there exists in DSD due to less aggressive filtering and potentially higher resolution is lost in this process? Not sure. I’ll try to convert to higher sampling rates to see if it makes a difference.

 

Personally, I’ve stopped listening to DSD primarily because it (subjectively) sounds a bit wrong to me on my DACs. To my ears it adds this tiny level of brittleness to the sound that makes human voice and strings sound  bit more natural, but makes a lot of instruments sound less ‘whole’ — don’t know if I can explain this better. Very minor effect, but enough for me to notice. Of course, on the objective side, the lack of processing tools in DSD and large file size for high DSD rates is also a problem for that format.

 

Ok, good to know. The piece of the puzzle that was missing for me was the large amount of quantization distortion in the audible range.  I can understand the need to filter it. 

 

Thanks for the info!!