Return-to-zero shift register FIRDAC

BTW, these scope shots are taken at the dac, but right after the DC blocking caps.
Here are a couple with the scope set to hi-res mode:
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Mark,

I don´t understand these images when comparing them to my measurements when playing a silent track at DSD256.
See spectrum in the attachment up to 1Mhz, but after Marcel´s analog reconstruction filter.
In your case without this filter, I would expect HF to be significantly more prominent.

Hans
 

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  • DSD256.jpg
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Perhaps it would be if measured before the DC blocking caps, and or if all cable and transformer loading on the dac output was removed.

What I don't quite get is why we should care? If the passive filtering as it affects the dac output is effective, then isn't that the goal?

Just keep looking for noise as a sanity check?
 
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Perhaps it would be if measured before the DC blocking caps, and or if all cable and transformer loading on the dac output was removed.

What I don't quite get is why we should care? If the passive filtering as it affects the dac output is effective, then isn't that the goal?

Just keep looking for noise as a sanity check?
Mark,
I just can’t imagine that the blocking cap etc can change the spectrum to that extend.
But there are two reasons to comment
1) when this spectrum is correct, it means that the digital signal offered to the JLSound’s board is already HF filtered and thus not being a true .dsf signal.
This digital filtering might have impact on the perceived sound reproduction
2) the more HF, the more likely it will interfere with the audio signal because of improper termination of your interlink.

Hans
 
...when this spectrum is correct, it means that the digital signal offered to the JLSound’s board is already HF filtered and thus not being a true .dsf signal.
Do you mean as configured? I mean if you have a discrete resistor dac and you put a big cap across the output then you expect to see some loss of HF at the dac output, right?

In the case of this dac it has a nominal output impedance roughly equal to the the positive phase resistors in parallel. If it were matched to a perfect filter then the filter would look resistive? If not perfectly matched then the passive filter might filter out some frequencies right at the dac output, roughly like putting a big cap there, possibly with some shelving.
 
Mark,

Yes of course, a resistor and a cap will attenuate HF as a 1th order filter.
But Marcel’s 4th order analog reco filter, starting with the same R-C, will even attenuate a lot more and not less :ROFLMAO:

So, when your spectrum is showing way less HF as what I measured in #1774 at the output of Marcel’s reco filter, it can only be because the software you are using, that’s offering a signal to the JLSounds board, is doing filtering in the digital domain.
That’s what I meant by saying that it’s no longer a straight but a filtered .dsf file.
And this digital filtering may affect sound perception and also the Frequency Response beyond 20Khz.

Hans
 
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...don’t know what Mark is using as a starting point, PCM or DSD?

Here are some pics playing real music using Simple DSD Converter.

Was I not clear?

The music sounds great. Cymbals sound in normal balance with bass, etc. If the FPGA-based Simple DSD Converter modulator can reproduce real music while not producing any HF noise then it would seem the guys who wrote the firmware must have solved one of the main problems with DSD. However, that seems rather unlikely.
 
Was I not clear?

The music sounds great. Cymbals sound in normal balance with bass, etc. If the FPGA-based Simple DSD Converter modulator can reproduce real music while not producing any HF noise then it would seem the guys who wrote the firmware must have solved one of the main problems with DSD. However, that seems rather unlikely.
Mark,

It wasn’t clear enough to me what these converters are doing.
When using JRiver to convert PCM to DSD, I get a straight .dsf file, but your converter is obviously doing more than that.
I already mentioned Marcel’s 8 bit PWM SDM.
This reduces HF noise significantly but it then deviates IMO from a DSD256 .dsf file.

Hans
 
Real music is non-PSS. Cymbals don't play constantly nor at a fixed signal level. There is no reason they should show up as a strong spectral line in an FFT?

Besides, the exact same music made from the same wav file sounds pretty close using Simple DSD Converter as it does using two different software based converters. Also have the vinyl of the same music. Mix balance levels of bass and treble instruments are very close to the same no matter how the music is reproduced. Reproduction of sound stage and space is a little different, but the differences are not gross.

EDIT: I will go take a look at it again later to see if I can find any problems. Maybe a problem with how the scope is calculating the FFT, or maybe some other issue not clear at the moment.
 
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