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Old 23rd February 2015, 10:32 PM   #201
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Originally Posted by hop.sing View Post
Well, setup is: Dam dac into adc burr brown pcm 4222 evaluation board, which is the master clock also, all connected via spdif with a rme firefac ufx, monitoring dac is a gustard x10. Monitored either through Geithain 904 or Joachim Gerhard Kalasan Speaker. The reference track has just one dac through the gustard x10, the processed tracks have one dac and adc on top of it. I can hear filter differences pretty well, the pcm4222 is not totally transparent, but good enough.
There are still variables, but that is the best testing method for me I can think of, since I want to use the Dam in the studio in precisely that application.
Tobias
Thanks. It wasn't clear what your reference for comparison was, and that answers my question.

Obviously listening to DAM -> PCM4222 -> FireFace -> Gustard x10 (ESS9018 based) -> amp -> Speakers is going to be a very different experience to Fire Face -> spdif -> DAM -> amp -> Speakers.

The main issue (apart from playing back a recording of the DAM via a Sigma Delta DAC) is that the filtering is additive, so you are listening to the effects of a filter which is the convolution of DAM + PCM4222 + Gustard x10 filters. This means that when you say "closest to the source" you are effectively selecting the filters that have the least influence on sound when combined with the filters in the PCM4222 and x10.

If that is the exact setup up you are planning on using it's entirely valid way of comparing, but it will not be an accurate reflection of how the filters behave with the DAM as a standalone.

cheers
Paul
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Old 23rd February 2015, 11:06 PM   #202
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Interesting point that hop.sing makes about referencing to a source, but I'm not sure that comparing to another DAC as all DACs add some colouration to the sound.

About the best I can think of is the approach used in one of the dCS papers which is to use an analog source as reference. About the only way I think I can feasibly do this is:

TT -> phono stage -> preamp -> amp -> speakers
TT -> phono stage -> ADC -> spdif -> DAM -> preamp -> amp -> speakers

The unavoidable downside is that it introduces an additional conversion step in the form of the ADC, but gives you a non-digitised source which can be used as a reference. It shouldn't matter that the TT/phonostage has it's own coloration as what is at issue is the ability of the ADC/DAC to accurately reproduce the signal it is fed.

cheers
Paul
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Old 24th February 2015, 06:17 AM   #203
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Originally Posted by soekris View Post
The FIR2 filter will also be modified to be pretty soft....
I will concentrate on NOS "filter" at first upsampling via PC to get the basic impression of the DAC. What do you mean with pretty soft? A slow filter? No filter could be very interesting too. Deemphasis - Are you detecting the use in the bitstream and applying the filter in this case only?
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Old 24th February 2015, 07:15 AM   #204
zfe is offline zfe
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Originally Posted by soekris View Post
Just so my plans are known:
...
There will come an official soekris release, all done using rePhase, but it take a little time to make and test all those filters.... Keep working on them here as it give me a lot of input what to do....
I would apreciate very much an intermediade firmware release (comming soon) only fixing the sample-rate-change clicks and power-off plops.
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Old 24th February 2015, 09:28 AM   #205
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Originally Posted by leehan View Post
Reduces ringing length and amplitude as well... hmm. We need Pos to to explain why he recommended intermediate centering values for IPs. Pos, where are you?
Hello leehan

I don't think I recommended any specific centering for the impulse per se.

The amplitude/phase relation of the target will dictate the way ringing is distributed around the impulse peak (pre/post).
Windowing (centering+number of taps+windowing algorithm) should be chosen to accommodate that target as much as possible.
Energy centering "should" give the best result, but you can also play with percentage values (especially if you know the pre/post ringing behavior).

Of course you can remove some pre ringing by forcing a small centering value (in effect truncating it), but this is probably not the best method -not the intended method at least- and it will alter your result curve and amplitude/phase relation.

If you use minimum-phase filters you will get no pre ringing and only post ringing, and if you use linear-phase filters you will get both pre and post ringing.
So by mixing minimum and linear phase filters, possibly at different frequencies and with different slopes and shapes, you can somewhat "distribute" the ringing around the impulse peak and choose how much pre and post ringing you want, and at what frequency it will appear.
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Old 24th February 2015, 09:40 AM   #206
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Originally Posted by spzzzzkt View Post
Thanks. It wasn't clear what your reference for comparison was, and that answers my question.

Obviously listening to DAM -> PCM4222 -> FireFace -> Gustard x10 (ESS9018 based) -> amp -> Speakers is going to be a very different experience to Fire Face -> spdif -> DAM -> amp -> Speakers.

The main issue (apart from playing back a recording of the DAM via a Sigma Delta DAC) is that the filtering is additive, so you are listening to the effects of a filter which is the convolution of DAM + PCM4222 + Gustard x10 filters. This means that when you say "closest to the source" you are effectively selecting the filters that have the least influence on sound when combined with the filters in the PCM4222 and x10.

If that is the exact setup up you are planning on using it's entirely valid way of comparing, but it will not be an accurate reflection of how the filters behave with the DAM as a standalone.

cheers
Paul
Yes, these are the Problems I am facing with this method, but still, The Gustard is on both files, so the variable is the pcm4222 ADC.... Of course, the filters build up, so results get blurry.
The closest to the source question with converters is so hard to answer... Because all files are at least converted once, probably even more often.
So I think, if this is going to be your Listening DAC, it is completely valid too use a filter that sounds best and not closest to the digital source.
Thinking about it... since all digital files are already butchered with at least one conversion, NOS might even be closer to the analog source, because it might restore some dynamics, that got lost in ADC. Just thinking loud....
Quote:
TT -> phono stage -> preamp -> amp -> speakers
TT -> phono stage -> ADC -> spdif -> DAM -> preamp -> amp -> speakers

The unavoidable downside is that it introduces an additional conversion step in the form of the ADC, but gives you a non-digitised source which can be used as a reference. It shouldn't matter that the TT/phonostage has it's own coloration as what is at issue is the ability of the ADC/DAC to accurately reproduce the signal it is fed.
But you really would need the very best ADC, but than that could work well, but only with records that never saw a conversion filter before :-)
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Old 24th February 2015, 12:51 PM   #207
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Volume matching between files is extremely important... 0.1 dB louder translates to better most of the time...
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Old 24th February 2015, 08:17 PM   #208
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Originally Posted by hop.sing View Post
But you really would need the very best ADC, but than that could work well, but only with records that never saw a conversion filter before :-)
That is true - the TC isn't bad for what it is, but . I also have a Sound Devices 722 which has excellent preamps and fairly good convertors. John Marks did a review of sorts in Stereophile which put it just short of a Grace 201/Tascam DV-RA1000 recording in DSD mode. This was back in 2006 however.

It shouldn't matter if there are conversion filters if they are in the source - just as you can't control the filtering done when producing CD's or digital downloads. Anyway I'm sure I can find a pre digital pressing or two - I have very little vinyl pressed after 1990, I ain't a new skool vinyl hipster

The issue I was pointing in your configuration is that DAM, ADC and x10 all ADD filtering to the original source. Anyway I think we'll have to agree to disagree on the validity of your testing.
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Old 24th February 2015, 08:51 PM   #209
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Originally Posted by pos View Post
If you use minimum-phase filters you will get no pre ringing and only post ringing, and if you use linear-phase filters you will get both pre and post ringing.
So by mixing minimum and linear phase filters, possibly at different frequencies and with different slopes and shapes, you can somewhat "distribute" the ringing around the impulse peak and choose how much pre and post ringing you want, and at what frequency it will appear.
I found that if you use a linear filter with adjustable slope it's reasonably straight forward to control the pre/post ringing by tweaking the slope of both Linear and Minimum phase filters.

There is an AES paper from 1995 written by Heylen and Hawksford which presents an algorithm for continuously adjustable phase in FIR filters. The paper includes a supposedly complete but barebones C++ implementation, but after entering the code I've discovered there is at least one macro missing for a critical step, and a second macro that is not used in any of the code. The two macros have different numbers of parameters so it's not a case of one being misnamed...
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Old 24th February 2015, 08:53 PM   #210
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Originally Posted by hop.sing View Post
P.S.
Volume matching between files is extremely important... 0.1 dB louder translates to better most of the time...
There are volume differences between many of the filters, so you'll need to recalibrate on a per filter basis.
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