Thanks a lot Fredrik, your explanations and recommendations are very appreciated.
Since most of the music I listen to is uncompressed FLAC 16-44 from my home network (Synology NAS) and some 24-96, I think I’ll stick to DXD, filter 1 or 6. I usually favor details and crystalline sound over a more rounded one as long as it’s not harsh or agressive.
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For almost a year of owning Bartok, I used DXD upsampling with filters 3-5-6, now I use DSD upsampling with filter 4. At first it seemed to me that I did not hear the difference between them at all))
I don’t have much DSD content and I’m not really worried about that. But the filter for DSD is always set to 1
I’m a bit confused.
You say "
A few words later you say
Hi Pete, in the latter case, I meant 1st of 4 filters for DSD content, not for PCM
I’m a bit torn because the DXD upsampling has clearly more resolution than the DSD, but my system benefits from the softer treble of DSD. I’m coming from a PS Audio Direcstream DAC, which is a different sound, but I do miss the DSD treble a bit.
I’m hoping someday the DSD upsampler goes up to DSD128 or 256 to provide a more equivalent resolution and dynamic range to the DXD. I suspect this is somewhat tied to the Bartok’s output modulator (single rate DSD sample rate, but at 5 bit), which could mean we are stuck at single rate DSD upsampling. But then DSD128 files sound fantastic on the Bartok…
I think Rossini can upsample to DSDx2. Is that to coax us switching over from Bartok? 
Bartok supports the same.
As of right now, it cannot.
Bartok can stream dsdx2 but not an option to upsample as such.
I’m struggling with these filters: I can’t hear any difference between any of them. Does the change take effect immediately on selecting the filter in the app or do you have to wait or do something else to activate the selection? Or am I deaf?
The filters are not effects. They are pretty subtle in what they do. I can only say that as you become increasingly used to whichever dCS component you have then their individual characteristics will become clearer. Even though I may not be able to directly hear the difference ( as if e.g. turning up a bass control) there are some combinations of input and filter that I find quite unnatural and to be avoided. Mind you I have been doing this for over 20 years.
I expect that in the end you will select one as being suitable for you and your system for each resolution and/or input. Just leave it at that and worry about it no longer.
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I notice that they fade in over a second or so. The easiest ones to hear are the minimum phase filters (Filter 5 at 44.1 and 48khz, and Filter 6 at 176.4, 192, 352.4, 384khz).
The things to listen for as you cycle through the filters are the level of punch or impact, how sounds move forward or back slightly, and how the top end changes slightly. What you should find is that the higher the number filters (for instance 3,4) have more punchiness and speed, are more forward, but are slightly more edgy on top. With the minimum phase filters you get the most focused impact and well-defined edges. Filters 1-2 have the most clean, polite, and balanced sound.
Or, as I would put it ; No.
The choice of filters is personal and depends upon not only your hearing and expectations but on the system components that you have, your room and the music that you typically listen to. For my part I would find your recommendations to have almost the opposite character to those that you ascribe to them.
In any case filter numbers do not characterise the filters as these change in accordance with the input format. So filter 1 is not the same for 16/44.1 and DSD 64. Or filter 5 is not the same type of filter for 16/44.1 ( asymmetric) and 24/176.4 ( Gaussian or approximate to it).
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Well, it is fair not to create expectations when evaluating this, and there is always the possibility that a filter setting does the exact opposite of what you would expect in your system.
But factually: PCM filters 1-4 are more or less the same at all samples rates, and progressively trade off alias suppression for transient speed. Hence, you get faster transients, and more alias distortion. To me that sounds like more punch and edge. And with minimum phase (Asymmetric) filters you get the fastest attack because there is no pre-ringing, but there is some small amount of phase shift. Again to me that sounds like a more forward and focused presentation with a fast attack, but I suppose it could always do the compete opposite for some totally random and mysterious reason.
Thank you very much for clarifying that. Was just concerned the Bartok filter was “stuck” for some reason and I was missing out on something amazing.
Changing filter on the Auralic Vega G2 did make a subtle but noticeable difference to midrange transparency particularly piano tones.
Good luck, these 51 year old cochleas can’t discriminate any differences at all. At least I can hear the difference crossfeed makes!
I seem to hear a difference between PCM filter 1 and 4, 5 and 6 now. Higher filter numbers seem to have little more liveliness about them. Quite subtle though. One of the biggest differences in how a live instrument sounds compared to a recording is in the transients. Now whether that is down to filters, sample rates or dynamic range compression of the transient peaks will depend on the recording. Any way to preserve those transients is to be welcomed.
Would anyone like to stick their necks out and describe what “(unwanted) Nyquist images” sound like?
If you haven’t seen it and FWIW here’s a post from dCS regarding filter choice
No idea what “(unwanted) Nyquist images” sound like 
but I don’t want them
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A Scandinavian avant-garde jazz band perhaps? Jazz isn’t my thing either. 
Nice link thank you.
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Thanks a lot. All of you teach me how to listen to the different filters. Especially @PAR and @IanB52. I am aware of the different listening experience you both heard. And riding on my own experience, I understand why people always said it is personal preference and depends on listening environment.
My experience similar to IanB52, however I would describe in another way. The audio I am listening are in 44.1KHz. So F5 is asymmetrical, no pre-ring but higher after ringing.
For me, the different between F1 to F4 is F1 is more subtle and F4 is playing more hard. Imagine a sax player, when he play hard, he will blow a short but strong and fast air into the instrument. You will hear the sound is soft at the very beginning but suddenly it will reach the max volume. You can feel the player is in high mood, the sound is vibrant. (That match most people say it is punch and faster). As the volume of that note come quicker to the max volume, it give me a feeling that it is closer and more focus. This is what I got from F4.
F1 give me a relatively opposite listening experience. It is more like the sax player try to play at night. He doesn’t play so hard, so the volume raises slowly and the max volume comes later. As it is softer, my brain imagine he is standing a bit away. Though it is subtle, that how my brain intercept the signal.
F5 is even faster and come closer. However the after-ring kill me. I got an inception of echo sound. Its like the sound need more time to disappear (due to longer after ring). Because my listening environment is small and closer to the speaker, I want the sound disappear quicker so that it won’t affect the note coming after. This is much easier to experience with kick drum in those jazz record.
F6 is even softer than F1, IMO. If I am listening in a larger room, I would prefer F4 or even F5 cause the room give the time to digest the punch and it is more vibrant. Now I am using F1. It give me a more comfortable feeling in a smaller room. Like listening in night time. My room can digest the sound more, make me feel clear and less chaos, especially in the bass.
So my conclusion is old school: it is a personal preference and depends on your listening environment and type of music.
My two cents sharing.
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