jt25741

Can anyone share their experiences on how the different Jitter Eliminator settings influences the sound of their system? I have been experimenting and find that Maximum produces a more mellow -- perhaps even less fatiguing sound. But I want to hear back to see what other owners of the DAC2 DSD-SE V2 feel about it. I am wondering what is happening to the signal to actually change the sound quality, save for reducing jitter. I would expect jitter to already be low on the USB interface, for which I use. Also, as much as I like the Apodizing filter naturalness --- it is not nearly as open/spacious and detailed sounding as the SLOW filters. The minimum phase filters are unlistenable to me, bright and unnatural/digital sounding. It would be nice to have a SLOW roll-off Apodizing filter...combining the best of both worlds. But in looking at the frequency response of the filter in the docs, there is too much ultra-high frequency noise that has to be suppressed to pull off a SLOW-rolloff filtering using the built-in filters (just my speculation why it is not available from ESS). Appreciate everyone's experience, who own the DSD-SE V2 version. Thanks

Unfortunately this is not correct either. The ES9016 is a much cheaper part. Also the ES9018 completely allows the bypassing of internal filtering. The most popular and acclaimed currently is the Auralic Vega. All filtering is done externally there and is a "selling point". I spoke with the designer on the Vega a few months ago, and he has some rather strong opinions on the 9018 filtering too In response to my question "I am curious if you are familiar with other ESS 9018 Designs that have an explicit analog filter stage, and can identify one or more. Just for my curiosity." You said: But now you corrected that statement, and it makes much more sense to me (thank you) but not entirely because of your comments on the 9016/external filter. As you replied in my earlier posts, this was my original position too, so here were are in apparent complete agreement. So back to the original problem at hand, I am not questioning what you hear and the need for extra upstream filtering. But logically, the way DACs work, in particular delta sigma DACs, is significant ultrasonic quantization noise (this is distortion) is introduced in the process. This has nothing to do with the filtering you do on the PC/MAC, as it happens later downstream on the DAC. This is what needs to be removed -- you cannot filter it out before it gets inserted in the audio path. Things like clock noise will get removed by the fully differential output circuit as these are common mode in nature --- and will simply cancel out with designs like the W4S DAC2. There are lots of reasons in the W4S you do not need external 2ndary explicit analog filters, and this is one of the reasons why it sounds so good. I am really happy you agree with most of us that this unit is in a class by itself among most if not all DACs within reason of its pricerange... It is a pretty standard reference design for an ESS Sabre.....but implemented with care and parts selection that push it far up in performance. DSD support is a 2ndary feature of the ESS Sabre --- and based on the operational complexities of the W4S DAC and IIR filter choice (this is for DSD only), it is much more basic/limited IMO in that regard. Said another way, and more directly, I would not buy the DSDse for DSD playback as my primary goal -- rather redbook and other PCM formats. DSD is an added feature that can sound good and comes along for the ride.

I thought the Ayre QB-9 was a ESS9016 design ,not ES9018. The slow filter is hardly wide-open...it is -100db down at stopband. It also has better group delay numbers. Stereophile tests all kinds of configurations often....but I have not seen W4S lately in their pubs. But often show fast/slow filter plots. I agree it sounds great.

Beweaveit, What I mean by reboot is that the IIR bandwidth setting (50Khz is default -- which happens to be default recommendation for Scarletbook SACD decode) used for DSD playback can only bet set on the unit by going into menu mode, changing to setting to another choice, and resetting (which is a reboot of the DAC). Doing this everytime one moves from DSD64 to DSD128 is kinda difficult, especially for those of us who put the DAC in a cabinet somewhere. Does this make better sense now? It is interesting that I hear issues with DSD and you with PCM...you put us together and the unit is perfect I know the unit sounds quite a bit better than others I compare it to for harmonic correctness and fatigue-free listening with PCM data --- which is only using digital filtering with the SLOW setting for me. With this said, I have no doubt what you hear is real, and perhaps you are more discerning with the lack of an actual analog filter on the output stage. An interesting data point will be to see spectra from the unit on a bench to see what in-band artifacts escape the digital filter and contaminate the integrity of the signal with various input frequencies. Most of us will have to wait until a publication like Stereophile gets a hold of one to put it under test for that. Until then, we have our ears which is the final judge that matters. I am curious if you are familiar with other ESS 9018 Designs that have an explicit analog filter stage, and can identify one or more. Just for my curiosity. As far as the high-bit rate sync issues....sorry....I cannot be of any help --- but based on your finding...I suspect I would have similar trouble......if not everyone with similar content.

Beweaveit thanks for sharing this. I have not attempted these higher PCM sample rates, but it is troubling to me that you are having such sync issues. Have you already pursued the path of calling W4S tech support to work with them on this issue? I have not called them for anything yet, but hopefully they will provide some solutions or root cause at minimum. Please share what you have learned on this if possible. As far as your other comments, my listening has found the "slow" rolloff far superior sounding.... but have not felt the need for additional upstream filtering as you have. My understanding is through the upsampling process, the nyquist frequency is pushed so high that the natural bandpass of the analog output circuit is sufficient. Have you compared "slow" with no filter with "slow" and the filter? For DSD64, quantization noise is an issue for my system and ears, and anything higher than default 50Khz is objectionable and suboptimal. I think this unit will have disadvantages over more complex/elaborate DSD decoding schemes involving upsampling and much higher frequency roll-off. I have not listened to DSD128, but filtering should be moved way up (reboot sequence needed on the DSDse) -- so switching between DSD64/DSD128 and getting optimal sound is not operationally easy.