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Old 24th February 2015, 10:35 PM   #211
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Quote:
Originally Posted by spzzzzkt View Post
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.
The first conversion (and a record will have two conversions at least, if recorded digitally) is the most destructive one, in my experience, so you probably will hear more differences with true analogue soundfiles.
Yes, my testing is seriously flawed, but has definitely advantages (together with disadvantages) over subjective listening in an uncontrolled environment, but through a shorter signal path, you can't deny that.
Quote:
Originally Posted by spzzzzkt View Post
There are volume differences between many of the filters, so you'll need to recalibrate on a per filter basis.
That's what I meant :-)
Tobias
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Old 25th February 2015, 08:37 AM   #212
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Quote:
Originally Posted by leehan View Post
Strangely my dac doesn't lock with MB2b.skr...?
Yes, quite odd indeed. Doesnít work on my dac either. We ARE talkng about the same dac board, arenít we? Mine is 0.01% FW rev. 0.9.
Is there a way that uManager can confirm the actual filter?
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Old 25th February 2015, 08:42 AM   #213
leehan is offline leehan  United Kingdom
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Hi Pos, thanks for your explanation.

Quote:
Originally Posted by pos View Post

I don't think I recommended any specific centering for the impulse per se.
Not a specific one, but I interpreted your suggestion in this post (Reference DAC Module - Discrete R-2R Sign Magnitude 24 bit 384 Khz) as using 1% for MP, 50% for LP and any value intermediate values for IP.

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Originally Posted by pos View Post

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).
May be I avoided "energy" centering as it sounded a bit mysterious to me Does it correspond to a particular percentage value or is it a different algorithm (sounds like a dynamic/variable thing)? And yes, my intention at that moment was to achieve a 30%/70% distribution of pre/post ringing respectively.

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Originally Posted by pos View Post

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.
What would be the difference between these 3?

A. Minimum phase filter with centering = middle
B. Minimum phase filter with centering = 1%
C. Minimum phase filter with centering = energy

I expect at least 2 of them to be the same or very close.

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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.
Ok, LP and MP parameters are the main tools for adjusting pre/post ringing ratio, not centering.

Cheers Pos!
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Old 25th February 2015, 08:47 AM   #214
leehan is offline leehan  United Kingdom
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Quote:
Originally Posted by SebastianL View Post
We ARE talkng about the same dac board, arenít we? Mine is 0.01% FW rev. 0.9.
Is there a way that uManager can confirm the actual filter?
0.02% FW rev 0.9, so the same board.

None documented, AFAIK.
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Old 25th February 2015, 08:47 AM   #215
TNT is offline TNT  Sweden
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I will reload what I uploaded to check if there has been some problem on the way somehow. I'm on 0.01% FW rev. 0.9.

//
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Old 25th February 2015, 09:39 AM   #216
pos is offline pos  Europe
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Quote:
Originally Posted by leehan View Post
Not a specific one, but I interpreted your suggestion in this post (Reference DAC Module - Discrete R-2R Sign Magnitude 24 bit 384 Khz) as using 1% for MP, 50% for LP and any value intermediate values for IP.
Yes, 1% if your target curve is MP, not to turn a filter into MP (even if it can work, in a way).

Quote:
May be I avoided "energy" centering as it sounded a bit mysterious to me Does it correspond to a particular percentage value or is it a different algorithm (sounds like a dynamic/variable thing)?
Energy centering tries to find the best window position (ie centering) around the impulse peak to maximize the ringing energy inside it.
For example it *should* return something close to 0 for MP targets, and taps/2 (middle) for LP targets.
For MP targets It does not work as intended most of the time, for some reason that I'll have to investigate...
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Old 25th February 2015, 09:53 AM   #217
leehan is offline leehan  United Kingdom
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Quote:
Originally Posted by pos View Post
Yes, 1% if your target curve is MP, not to turn a filter into MP (even if it can work, in a way).


Energy centering tries to find the best window position (ie centering) around the impulse peak to maximize the ringing energy inside it.
For example it *should* return something close to 0 for MP targets, and taps/2 (middle) for LP targets.
For MP targets It does not work as intended most of the time, for some reason that I'll have to investigate...
So the ideal method of developing filters would be:

1. Middle centering for LP
2. Try both energy and manual centering for IP (intended ratio) and MP (1%), and test which one performs better.

Many thanks Pos, both for your explanations and for developing rePhase of course
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Old 25th February 2015, 10:02 AM   #218
leehan is offline leehan  United Kingdom
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After listening to a few hundred albums of various genres, here is an interim impression report:

There is a BIG difference from recording to recording. So I'd suggest reviewers to listen to many, not just your most favourite one.

Of course it's important to make our favourite recordings sound good. Then, we are in the excitement territory rather than being true to the source. So I'm glad that we'll have multiple filter options. As long as each filter is developed properly, all will have a use.

For the record, my set up is the most basic: Amanero USB > dam1021 > single ended buffered output > Beyerdynamic DT880 600ohms. This way I can eliminate contribution of amplifiers and crossovers in speakers.

Dam is powered by a transformer. It turned out that buffered output impedance and voltage level I use matches well with my headphoes for low distortion.

[Off topic blurb]
This setup should be scoring really high on price/performance in our hobby of diminishing returns.

To whet some more appetite for dam1021, with those good sounding recordings, I can easily figure out the distance between the mic and the acoustic instrument, whether a guitarist is using his nails or fingertips to play the guitar, and the fretbuzz of electric basses that I haven't heard or paid attention before (up to the point of being slightly disturbing ). Not to mention the realism of the instruments. This is new to me, so I'm excited. Others who are already used to this quality may not be so. Bad studio setup or mastering is equally obvious on some others.
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Old 27th February 2015, 05:12 AM   #219
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Default How about using HQ Player to upsaple to 384 Khz ?

I have been deep into DSD 256 and HQ Player the last 8 months and so have completely missed out on this exciting R2R arrival.

During my explorations with HQ Player (available for Win, Mac and Linux), I have found that the upsampling of 16/44 to 24/352 or 384 is done very well. There are lots of choices for filter types tochoose from. It will all run on a i5 no problem for PCM upsampling. HQ Player also has a NAA = Network Audio Adapter which was only available for Linux, but now is available for Win and Mac as well, so no problems with USB drivers. The NAA can run on a separate computer, and needs only to be seeing a common DHCP on the LAN as the HQ Player Desktop machine. We have also found that the NAA PC can be optical isolated on the LAN using a pair of inexpensive Fibre Media Converters which has a very big effect on shutting out noise coming over the CAT 5 (although you have to be careful about the power supply going to the FMC attached to the NAA PC)

So by using HQ Player to do the umpsapling to 352 or 384, you would only need to have a nice 352 or 384 filter, a much easier task than a filter that does a good job on 16/44

Apologies in advance if I am repeating a previous post, I have not read the whole thread
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Old 27th February 2015, 08:00 AM   #220
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Quote:
Originally Posted by EuroDriver View Post
So by using HQ Player to do the umpsapling to 352 or 384, you would only need to have a nice 352 or 384 filter, a much easier task than a filter that does a good job on 16/44
Regardless of where it is done you have to deal with the imaged generated by resampling at some point.
Any kind of upsampling or oversampling of 44.1kHz still has a 2.05kHz transition band between the audio band and fs/2.

The DAM basically does in the FPGA what HQ Player does in software.

Last edited by spzzzzkt; 27th February 2015 at 08:05 AM.
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