Filter brewing for the Soekris R2R

Hello, Alter having fiddle a lot with the various filters available for the Soekris R2R, I find that the best is just to not take care of filters at all and upsample to 384 Khz from my software (ROON).
Am I the only one satisfied by this approach and is there a downside I did not think of?
Thanks for sharing your thoughts.
 
Hello, Alter having fiddle a lot with the various filters available for the Soekris R2R, I find that the best is just to not take care of filters at all and upsample to 384 Khz from my software (ROON).
Am I the only one satisfied by this approach and is there a downside I did not think of?
Thanks for sharing your thoughts.

Also using Roon, I haven't got to there yet. I have played with the filters in Roon, and upsample like you, but I have for now chosen to let the Soekris do the work. May I ask, do you find the filters and filter options in Roon better/giving another sound than the original Soekris filters/ the brewed?
 
Also using Roon, I haven't got to there yet. I have played with the filters in Roon, and upsample like you, but I have for now chosen to let the Soekris do the work. May I ask, do you find the filters and filter options in Roon better/giving another sound than the original Soekris filters/ the brewed?

My understanding is that if you send a signal of 384Khz to the DAM it will bypass the filter. Ever since I upsample from Roon, I find that the sound is more clear and ethereal. From then I lost all interest in playing with filters.
As for the filters in Roon, I did not find much difference between them, but I believe they serve a complete different purpose than the ones on the DAM. I leave it on the default one.
Would be nice if you test and report back....
 
"I believe they serve a complete different purpose than the ones on the DAM"
I do not understand why it could be different purposes? I will try your suggestion, and check for differences, but filtering is filtering anyway, it is just a question where it takes place, and of course what king of filter, isn't it?
 
I'm working on a dams/dacs firmware upgrade which will include a doubling of filter coefficients, with to goal to further improve sound quality on all soekris dams and dacs.... A doubling would then be 4K taps for the 44K/48K PCM filter.

So I would like to restart this filter brewing thread, I really would love some feedback and help on the best FIR filter design techniques....

For the stock filters I used rePhase for design, a very nice and easy to use tool, but I don't know if it's optimal for DAC filter design. Are there better tools available that don't require a degree in math and cost a fortune ?

All inputs are welcome, but for it to be useful when creating a new set of stock filters I need to be able to recreate the process of generating the filters.

For breakthrough assistance I would give out a dac1541....
 

TNT

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Joined 2003
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Nice!

I hope that the winning contribution is "Nyquist" clean - i.e. no fancy "new ideas" on how the sampling theorem works etc.

For this to be a meaningful exercise, one have to make up one mind on some basic requirements on the end result (bat friendly or human oriented) or maybe a priority list of desirable properties. For example:

1st: aliasing suppression
2nd: band pass ripple
3rd: phase correctness
4th: inter-over safe
5th: original zero-crossing preservation
6th: upper frequency respons
7th...?

of course taking all 4k taps into play.

Improving exiting sold products often gives a lot of cred and happy customers, willing to do more investments as they seem to be future safe.

//
 
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Soren, have you tried the different Albretch windows in rephase?

No, I have no glue what they are or their advantages.... But I would be very interesting in information about what windowing create the best sounding filters.... I can see doing 4K taps make it easier to create more perfect filters.

You see, I don't have the time to create all combinations of filters and then listen to them with different music, headphones and speakers.... And I'm not sure how goo my ears are, or need more opinions....

The goal is to create a couple of new filters and then ask for feedback, but start out with as good as possible.
 
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At what sampling frequency do these filters run?
If you send me your current rephase setting files I might see if things can be improved in term of rephase parameters.

Incoming sample rate -> 352K / 384K. But it's not a question about correct settings, I have that under control, especially with the new 4K taps for 44K incoming sample rate. It's a question about what sounds the best, what kind of filters will give the best sound. And I know things like that softer filters often gives better sound.
 
but soren there is one more thing you need to consider is to have external filter selection system. So that consider we can select the filter type based on the button selection in the front panel like 8 filters. That will make easier to load and unload the filters. If you can let us know how to do it that will be really helpful.
 

TNT

Member
Joined 2003
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Hi Soren,
maybe this can help a bit: archimago about filters

Yepp!

"The key here is to remember that within a properly bandwidth limited signal where all the frequencies are below Nyquist, a linear phase FIR filter actually does not create ringing regardless of the impulse response appearance. As I have said in the previous weeks, any decent recording will follow this rule. And if it does, then the ideal filter to use is clearly a linear phase, sharp filter that can reconstruct all the frequencies in the audio data with essentially ideal temporal resolution.

And that folks is the "myth" we need to say goodbye to in 2018! Linear phase, steep "brick wall" type antialiasing/anti-imaging digital filters performed with high precision, and with no intersample overloading, do not "ring" with good recordings that only contain "legal" frequencies below Nyquist. Sure, some people might prefer minimum phase slow roll-off filters because they sound different (as per Ayre's "Listen" filter), but technically if you care about time domain performance and frequency domain accuracy of 44.1kHz playback, you would go with a high precision, reasonably sharp, anti-imaging linear phase reconstruction filter (as per Rob Watts' video linked 2 weeks back)
."

//
 
Glad to know a new set of filter is being developed.

Thus far I kind of prefer the red filter, however for some material it can be a bit sharp (not sure how to describe it). Since my old age ears can’t hear above 15k, what I have done is use foobar resample-v to upsampling to 352k/384k and start the rolloff around 15k; and targeting about -140dB @ Nyquist freq; before feeding the dac1541. Thus effectively replacing FIR1 with an external filter. My ears really like this current setup.

Reason for taking this path is because I still struggle to understand filter brewing with rePhase.

Who knows, the new 4K tap capability might revive filter brewing for the soekris DACs.
 
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Hi Soren,
maybe this can help a bit: archimago about filters

Yes, I agree with the author.:) Digital recording is limited bandwidth system like the CD. You can't have 8kHz square wave on the CD system. Both sine wave and square wave(more than (22.05/3)kHz) are same on the CD because of the limited bandwidth. In other words, the 44.1kHz system can have less rise time than the 96kHz system(this is probably a well-accepted thing), which means the impulse response of the 96kHz system has less ringing than the 44.1kHz one.

You need to use limited bandwidth impulse when you measure digital system. True impulse, which means no bandwidth limited, is divide by zero. As long as the impulse is a limited one, linear phase filter doesn't make additional ringing. Minimum phase or others do make some additional one.