Filter brewing for the Soekris R2R

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A

arteom

Member
Joined 2016
arteom said:
Hello,

I seem to have gotten myself into a jam and not too sure how to get out, hoping someone here can help. As stated in a previous post, I tried to send the DAC older filter after upgrading to the latest firmware version. Now the DAC seems to be stuck. I only have SPDIF input, and when connected the device just returns "L044", if I power it down and back up it shows -

R1.06
I3
L000
F5
v+00
I3

but the UI manager does not load. I was hoping to get it to that point and then try to again update firmware, but no such luck.

I am fairly new to all this, have mostly been following guides posted. Am using PuTTY to communicate with the DAC. Anyone have any ideas on how I can get myself out of this and get the spdif input working again?

Thanks,
Click to expand...

nevermind, I got it to go into ui manager. not exactly sure how, just kept hammering at it.
 
G

Green77

Member
Joined 2007
Hi all!

I have downloaded the zip file in post #1311. When unzipped i get a file named 1021filt_1.05NOS.skr.
Is this the filter it's suppose to include?

(F4 linear ) Linear Phase filter, Soekris.
(F5 mixed ) EQHQ_lpbr_b4 Linear Phase filter, Paul.
(F6 minimum ) "quasiNOS" C128dp Linear Phase filter, Paul.
(F7 soft ) New NOS filter, Paul.

Because when i upload the filterfile to my dam1012 i get some NYQUSIT filters i guess, they aren't called the names that i paste...
What is it that i miss?

Best regards // Daniel
 
R

rickmcinnis

Member
Joined 2006
Arteom,

I have had this same experience EVERY TIME I change umanager. the first time put me in a panic, but as you found, it eventually worked.

The last time was when I installed the corrected filter pack from oneoclock but with a twist. I did the install with the DAC playing music - turned the DAC off and on after finishing and all was fine. I always leave the DAC on and had, also, left the computer I control the DAC with running BUT when I turned the computer off for the evening and turned it back on the next evening the no umanager problem occurred again.

No longer induces panic attacks, luckily. Well, I turned the DAC on and off numerous times, the same with the computer. Could not get it to recognize umanager with the +++ but when switching the DAC on and off, with the computer on (in TERA TERM) it would display jibberish when turned off. Did it many times, same jibberish every time so I turned the computer off and just listened, for that evening, as is.

The next evening umanager was available without any drama.

Very strange! But, at least, it is not fatal.

Written in case others experience this. Doesn't matter what you do, but it will return.
 
hollowman

hollowman

Banned
Joined 2007
I'm a bit late to this filter party, and I have only glanced at this thread.
Some Q's:

Are any of the software (FPGA/DSP) filters IIR?
Also, the old (1992?) Meitner "IDAT" DAC switched dynamically between IIR and FIR based on the type of signal going thru it: "The solution to this quandary is to use both types of filters. In the IDAT, a detector looks at the nature of the audio signal and directs the digital audio data to the filter best suited for that type of signal. Musical components that are fairly continuous in level with no transients are routed to the FIR filter; signals with steep leading edges are sent to the IIR filter. The two filters' outputs are then combined and output to the DACs. The musical signal is thus processed by the optimum filter type for that kind of signal."

Has anyone done something similar with the soekris?
 
hollowman

hollowman

Banned
Joined 2007
Classic Philips 4x filter?

I saw the H i F i D U I N O blog, with its extensive table of soekris filters, but I'm not sure whether/if any are similar to (emulate) classic Philips 4x architecture from the mid-1980s-early 1990s.

Back in Spring 1995, Stereophile briefly commented on this (re. square waves and DACs).
(I almost never have seen DAC reviews/reports with squarewave metrics, in Stereophile or elsewhere).

Squarewaves (by JA)
A processor's reproduction of a 1kHz, full-scale squarewave can be either boring or exciting. Because the CD system is limited to a maximum bandwidth of half the sampling rate--22kHz--it can't actually reproduce squarewaves. A squarewave can be shown by Fourier analysis to comprise a series of odd-order harmonics regularly dropping in amplitude with increasing frequency. For perfect reproduction of a 1kHz squarewave, therefore, we would need to be able to reproduce the 1kHz, 3kHz, 5kHz, 7kHz, etc., components, all the way to infinitely high frequency. However, as the CD system will not reproduce the harmonics above the 21st, at 21kHz, the 1kHz squarewave will not have a true square shape, but instead will look like fig.1. It looks as though there is overshoot and ringing before and after each transition from high to low and vice versa; in fact, what you see is what is termed Gibbs Phenomenon--the effect of omitting the high harmonics that would otherwise "square up" the waveform.
An externally hosted image should be here but it was not working when we last tested it.
Fig.1 Waveform of 1kHz squarewave restricted to 22kHz bandwidth.
Most of the waveform plots we publish show this completely natural overshoot and ringing; and in this sense, looking at the shape of a 0dBFS squarewave reveals nothing new about the product itself. However, while some digital filter chips, such as the Philips designs, exactly reproduce this kind of mathematically correct squarewave, others, such as the common Burr-Brown and NPC chips, chop off the tops and bottoms of the waveform (fig.2). As full-scale squarewaves don't exist in music, this will have no deleterious effect on sound quality. However, all things being equal, it does allow the designer of the digital filter to maximize its low-level mathematical accuracy.
An externally hosted image should be here but it was not working when we last tested it.
Fig.2 Waveform of 1kHz squarewave restricted to 22kHz bandwidth and reconstructed with NPC digital filter chip.
There is one class of digital processors and CD players, however, that performs quite differently on this test. If their digital filters have been optimized for time-domain performance--Wadia, Pioneer's Legato Linear--they don't suffer from the Gibbs Phenomenon "ringing," and more correctly reproduce all transient-type signals (fig.3). The price to be paid, however, is a premature rolloff at the top of the audioband. The Meitner IDAT even tries to get the best of all worlds by using a DSP-chip-based digital filter for continuous signals. This filter produces a normal squarewave shape, but switches to a filter algorithm optimized for transient performance on signals that are rich in transients.
An externally hosted image should be here but it was not working when we last tested it.
Click to expand...

Bottom line:

Those with the soekris DAC at their disposal, and proper lab gear, please analyze (and publish results, here in this thread) squarewave response.
 
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Z

zfe

Member
Joined 2014
hollowman said:
I saw the H i F i D U I N O blog, with its extensive table of soekris filters, but I'm not sure whether/if any are similar to (emulate) classic Philips 4x architecture from the mid-1980s-early 1990s.

Back in Spring 1995, Stereophile briefly commented on this (re. square waves and DACs).
(I almost never have seen DAC reviews/reports with squarewave metrics, in Stereophile or elsewhere).



Bottom line:

Those with the soekris DAC at their disposal, and proper lab gear, please analyze (and publish results, here in this thread) squarewave response.
Click to expand...

Just compute the respone of your filter(s) of choice with a square wave. The DAM will reproduce the computed result with more accurancy than you can see it on those scope pictures.
The only thing to be aware of is that any output above +-1 (=full scale) will be clipped to +-1.
 
hollowman

hollowman

Banned
Joined 2007
I don't have a DAM..

zfe said:
Just compute the respone of your filter(s) of choice with a square wave. The DAM will reproduce the computed result with more accurancy than you can see it on those scope pictures.
The only thing to be aware of is that any output above +-1 (=full scale) will be clipped to +-1.
Click to expand...
I don't have a DAM (soekris 1021) to play with ...
... my point was that one of you lucky DAM owners (with proper lab gear) would put it on the bench and post some results (scope traces of square waves)...
 
Z

zfe

Member
Joined 2014
hollowman said:
I don't have a DAM (soekris 1021) to play with ...
... my point was that one of you lucky DAM owners (with proper lab gear) would put it on the bench and post some results (scope traces of square waves)...
Click to expand...

... my point was ;) you don't need a DAM.
What you will see on the scope is a property of the filter not the DAM. You can compute it from filter coefficients.
 
bear

bear

Member
Joined 2002
bearlabs.com
zfe said:
Just compute the respone of your filter(s) of choice with a square wave. The DAM will reproduce the computed result with more accurancy than you can see it on those scope pictures.
The only thing to be aware of is that any output above +-1 (=full scale) will be clipped to +-1.
Click to expand...

zfe, my head is on fire - no I mean my brain is melting...

Trying to follow this thread and about 3 others, plus read the blogs!

Please explain what one would do in practice to "compute the response of your filters...with a square wave"?

I'd like to do that... to me the response in fig 3 of the recent post is the one that is likely to actually sound best. In my view a dB of droop at 20kHz vs. overshoot is small price to pay.

AND, at present time what are the top filters that are out there for direct download?? I've got a V2 coming for evaluation... so would like to give that a shot over the stock filters.

Personally I am looking for natural voices with sibilence that "resolves" as part of the voice - not a smear, clearly different sounds to things like cymbals, lack of specific "sonic character" to the overall sound - more to the source, and VG left-right/front-back soundfield - especially capable of making large (well recorded) choral pieces sound clear and unmuddled. And to keep going, trumpets that sound like real ones, not nasty ones, harpsichords that sound like instruments not disparate parts... having said all that, I listen to more rock and jazz than classical, but my PCM 63 based DAC now antique does a lot of that not all, and doesn't do hi-res at all.

You know, holy grail sound... :D
Relaxed, open, so you just want to listen to your whole library...!

Currently using Quad57s with a discrete stepped "L" attenuator, and a pair of DC coupled BEAR Labs SE Mosfet amps - very minimalist, clean.

_-_-
 
C

chriss0212

Member
Joined 2005
NOS mode

Hi all,

To be honest, I am a bit confused by what NOS means by SOEKRIS. Afaik NOS means: no filter, no oversampling. Don't know if I may understand the NOS filters for SOEKRIS right, but for me it sounds, that it is no filter but with oversampling.

What do I have to load if I don't want any filter and no oversampling?

Thanks and greetings

Christian
 
S

soekris

Member
Joined 2009
soekris.dk
Paid Member
T

t27049215

Member
Joined 2009
oneoclock said:
jimmy2004y said:
when using "quasiNOS", it can't lock the 192Khz signal.
But I'm using the FW Ver1.05, so I didn't know it's the filter cause that or the FW cause that.
Click to expand...
Thanks jimmy2004y it was wrong. This is corrected. Operates at all speeds.
Click to expand...

Late to the party :p
Weird... I get no sound when feeding 192khz to "quasiNOS" while stock filter works.
using V4 board with 1.06 firmware.
 
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