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Filter brewing for the Soekris R2R
Filter brewing for the Soekris R2R
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Old 30th March 2015, 10:21 AM   #521
Stixx is offline Stixx  Germany
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Paul, we all owe you for your relentless work in creating new filters to try!
And I haven't even tried a single one since I just finished my breadboard build of the DAM... Will try in the weeks to come.

On another note: have Blue - Solar Fields playing while I'm typing... nice ambient stuff. Have you tried Connect.Ohm - 9980?
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Old 30th March 2015, 01:15 PM   #522
oneoclock is offline oneoclock  Europe
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Quote:
Originally Posted by DEQ+TheEnd View Post
oneclock – thank you for posting you phasing measurement. One thing not clear to me. Does it have any meaning where on the timeline the impulse peak are? I understand its usually less desirable with pre-ringing than post. But I’m unsure if the placement of the peak in your measurement are something we can read something from?
No. Impulse peak time is simply the filter shape and the coefficients number. More coefficients greater delay. But I see 2 ms delay is no important.

What is interesting is the slope time length of ascent and descent. It would have been ideal if the graph soften the peaks and valleys but the program does not make.

And if some filter spoiling phase too and can tweak to improve like 1021SA2F2v1.

Last edited by oneoclock; 30th March 2015 at 01:25 PM.
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Old 30th March 2015, 11:09 PM   #523
spzzzzkt is offline spzzzzkt  Australia
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Quote:
Originally Posted by Stixx View Post
On another note: have Blue - Solar Fields playing while I'm typing... nice ambient stuff. Have you tried Connect.Ohm - 9980?
Yes I have. I've a bit of an Ultimae fan since I discovered their Fahrenheit Project releases in the mid-2000's.

On the filter R&D front:
I've done the filter co-efficents using the v3 settings but sweeping phase for Minimum (p=0) through to Linear (p=50) in increments of 5. v1-3 are all done using p=30. I'll get around to making up filter files for these later today.

I had a quick play around with Octave's FIR2 routine and found I can get very similar but non-identical results to what SoX produces using b= to set kaiser-window beta.
Attached Images
File Type: png SoXvsFIR2.png (26.5 KB, 335 views)

Last edited by spzzzzkt; 30th March 2015 at 11:23 PM.
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Old 31st March 2015, 03:12 AM   #524
spzzzzkt is offline spzzzzkt  Australia
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I was about to post up batch of v3 filters when I had a listen to the -p0 version...

I had a suspicion that the minimum and intermediate phase weren't quite as sexy as the cheers leaders would have you believe. You only have to have a peek at the step response to see what is going to happen.

Changing phase doesn't remove ringing from the filter, it simply shifts it. And there-in lies the rub. If you move all the pre-ringing energy to after the impulse you double the amplitude of the post ringing, and that means the overshoot doubles too. Barf.

This plot shows the step response of the filter coefficients.

Filter brewing for the Soekris R2R-impcomp-png

Orange is p0 (-M) , green p30 , and blue p50 (-L).

-p30 sounds ok using volume control, -p0 was just plain awful.

It would be possible to drop the gain on these filter but the output is going to end up very low.

I've attached the folder of filters but be aware that anything less that p25-30 is likely to distort badly.
Attached Images
File Type: png ImpComp.png (46.9 KB, 807 views)
Attached Files
File Type: zip PhaseTest.zip (195.1 KB, 24 views)

Last edited by spzzzzkt; 31st March 2015 at 03:28 AM.
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Old 31st March 2015, 10:50 AM   #525
DEQ+TheEnd is offline DEQ+TheEnd  Norway
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Default Controlling Phase with Sox on

Once again an excellent post Paul. I been meaning to test exactly what you just did there. But can’t for the life of me figure out how to do those impulse plots?

From the SoX manual we can read:
Code:
A phase response setting may be used to control the distribution of any 
transient echo between ‘pre’ and ‘post’: with minimum phase, there is 
no pre-echo but the longest post-echo; with linear phase, pre and post 
echo are in equal amounts (in signal terms, but not audibility terms); 
the intermediate phase setting attempts to find the best compromise 
by selecting a small length (and level) of pre-echo and a medium 
lengthed post-echo.

Minimum, intermediate, or linear phase response is selected using 
the −M, −I, or −L option; a custom phase response can be created 
with the −p option. Note that phase responses between ‘linear’
and ‘maximum’ (greater than 50) are rarely useful.
Is –p given in samples or milliseconds?
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Old 31st March 2015, 10:52 AM   #526
jaffar is offline jaffar  Russian Federation
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Quote:
Originally Posted by spzzzzkt View Post
Yes I have. I've a bit of an Ultimae fan since I discovered their Fahrenheit Project releases in the mid-2000's.
Speaking of Ultimae Records, the track "Alone" by Jaia from the Fahrenheit Five always comes to mind as the most deep, spacious and dimensional one from the entire series... Even on my plain old 1616m it sounds incredible.

Would be interesting to supply the most "dimensional" filter for that track, brewed by an Ultimae fan
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Old 31st March 2015, 11:42 AM   #527
AndrewCee is offline AndrewCee  United Kingdom
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Hi, Got a quick prototype up and running while awaiting remaining parts to arrive. Tried just about all the filters on here - thank you - having read all 50-something pages with interest. Likely will have a go myself at some point once I figure out how I can measure the performance of the filters with my Tek TDS2024. I prefer the 'Crap V2 filters', the original and the latest one posted a few hours back. In my mind they offer greater weight and scale to the sound, dynamics if you like, and I think the most transparent of the lot. My aim, simply to reporduce what is in the original mix without colouration or bias of any kind. Guess that's what we all look for, kind of....??

Thanks for all the brilliant work and helping me enjoy playing electronics & music once again!

Andrew
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Old 31st March 2015, 11:51 AM   #528
spzzzzkt is offline spzzzzkt  Australia
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Quote:
Originally Posted by DEQ+TheEnd View Post
Once again an excellent post Paul. I been meaning to test exactly what you just did there. But can’t for the life of me figure out how to do those impulse plots?



From the SoX manual we can read:

Code:
A phase response setting may be used to control the distribution of any 

transient echo between ‘pre’ and ‘post’: with minimum phase, there is 

no pre-echo but the longest post-echo; with linear phase, pre and post 

echo are in equal amounts (in signal terms, but not audibility terms); 

the intermediate phase setting attempts to find the best compromise 

by selecting a small length (and level) of pre-echo and a medium 

lengthed post-echo.



Minimum, intermediate, or linear phase response is selected using 

the −M, −I, or −L option; a custom phase response can be created 

with the −p option. Note that phase responses between ‘linear’

and ‘maximum’ (greater than 50) are rarely useful.


Is –p given in samples or milliseconds?
neither!
0 gives 100% post-ringing and 100 gives 100% pre-ringing. 50 gives an equal balance between pre and post ringing or your standard linear phase impulse.

The step response was done in fuzzmeasure.
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Old 31st March 2015, 01:55 PM   #529
DEQ+TheEnd is offline DEQ+TheEnd  Norway
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Ahh, gotcha we just shift the null-point. Make sense. What idiot other than me would order a filter with ringing?
Quote:
"Pretty please, can I please have a filter with 90uS pre-ringing please?"
“Sir, we feel so sorry for your missing brain you can have two in dual mono with all the ringing you would ever want and need to compliment your tinnitus.”
What files do you feed fuzzmeasure? I have REW and Holm installed. But they both laughs at me when I try to import files.
Holm can’t make sense of anything above 192kHz. In my mind we need to simulate with intended coefficient table?
I can have Holm do impulse plot with
Code:
sox --plot octave -r 44.1k -n output.wav synth 1 noise  sinc  -b15 -L -20.5k -t3800  > filterDM_Linear.wav
But I’m not sure if that 44.1k fs plot would equal to? edit: fs would of course, I ment impulse respons with 44.1 fs?
Code:
sox --plot octave -r 352.8k -n output.wav synth 1 noise  sinc  -b15 -L -20.5k -t3800  > filterDM_Linear.wav
Click the image to open in full size.

Is it possbile to do impulse plots with Octave by altering this code?
Code:
[h,w]=freqz(b,1,4096);
plot(176400*w/pi,20*log10(h))

Last edited by DEQ+TheEnd; 31st March 2015 at 02:01 PM.
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Old 1st April 2015, 01:00 AM   #530
ylingf is offline ylingf  United States
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Here's some comments from Mike Moffat @Schiit regarding the filter design of Yggdrasil DAC, which is claimed by those who heard it as the next best thing.

" That the DSP filter in the Yggy is closed form, and preserves all of the original samples is commonly known. The filter is also time domain optimized which means the phase info in the original samples are averaged in the time domain with the filter generated interpolated samples to for minimum phase shift as a function of frequency from DC to the percentage of nyquist - in our case .968. Time domain is well defined at DC - the playback device behaves as a window fan at DC - it either blows (in phase) or sucks (out). It is our time domain optimization that gives the uncanny sonic hologram that only Thetas and Yggys do. (It also allows the filter to disappear. Has to be heard to understand.) This is combined with a frequency domain optimization which does not otherwise affect the phase optimization. The .968 of nyquist also gives us a small advantage that none of the off-the shelf FIR filters provide: frequency response out to 21.344KHz, 42.688KHz, 85.3776KHz, and 170.5772KHz bandwidth for native 1,2,4, and 8x 44.1KHz SR multiple recordings - the 48KHz table is 23.232, 46.464, 92.868, and 185.856KHz respectively for 1,2,4, and 8x.

The Pacific Microsonics PM100 filter (HDCD) was a variation of all of the off-the shelf cookbook filters of the late 80's with the standard Frequency Domain successive approximation coefficient calculation found in the Burr-Brown, Phillips, Analog Devices, and are still used on the front ends of most delta-sigma DACs today. I cannot comment beyond that because I was not interested in a format that had already been condemned to stillbirth because of its required encode - exactly what's wrong with DSD. You have to get all of the studios on board. As I have said before, this leads to recordings like: The Folsom Prison Orchestra playing Regional Polkas of the Balkan States, or better yet, the folkloric Orkney Island Shepherds Cries of Ecstasy upon Summer Moonlight. The latter was so well recorded, I've heard, that the listener could hear the shepherds gently guiding the young kids' rear legs into their high topped boots as the objects of the rapturous ecstasy gently bleat. Those recordings are not for me - my musical tastes are quite ordinary - my products support the 99.999 per cent of recordings. Again, I digress. Since lower frequency wavelengths are measured in tens of feet, placement in image gets increasingly wrong as a function of decreasing frequency in non time domain optimized recordings - these keep the listener's ability to hear the venue - not to mention the sum of all of the phase errors in the microphones, mixing boards, eq, etc on the record side. An absolute phase switch is of little to no value in a stochastic time domain replay system."
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