RedFuneral

Thanks, I appreciate your effort in measuring this. It seems safe to conclude that I'm not preferring the sound of this because it's somehow purer in the time domain or otherwise. I've actually owned this DAC since April and have been using the standard flat filters up until this past week. I like the DAC but really have only enjoyed music in short bursts, that changed when I found this 'one weird trick.' Subjectively it sounds like a major system upgrade where I'm not fatiguing while listening anymore. It sounds effortless and out of this I'm able to better focus on the music and follow along with it, vocals sound more real & fast music more legible. I've spent most of my freetime in the past 48hrs A/Bing this NOS mode vs alternatives(Standard filters, Windows upsampling, HQP) and my impression remains consistent that this mode is doing something unique. Now I have to admit most of my music is of subpar recording quality(still it is lossless 16/44.1K) and over the past decade I've listened through real NOS DACs more often than not. Do you have any theories as to why I much prefer this sound? Could there be an additive coloration or distortion that's familiar to past gear I remember fondly? I wasn't able to recreate this impression with any of the filters in HQP(they're similar to the normal filters built into the Gustard.) It also isn't the EQ as I've tested this with music without significant content >5khz which sounds identical with EQ toggled on/off. I likewise don't think it's just a preference for reduced treble as I just switched from an Asgard 3 amp to a Drop THX specifically to attain a brighter/faster treble.

I wasn't aware of the X26's NOS mode. Annoyingly I cannot find an impulse measurement of the X16 to confirm if it's the same. I only have the frequency response measurement I found here: You can see the reason for my intense EQ(I intend to scale it back to taste, the negative gain required is at the extreme of what I can compensate for.) Do you know if the X26 is doing this in the same way as the X16? It seems to be an implementation developed by ESS & built into the chips but I couldn't find any documentation of its existence outside the 9068AS when I was searching last night. I'm curious why the AKM super-slow approach doesn't sound NOS to my ears despite similar measured performance while the Gustard subjectively does. I don't have any experience to comment on other chips implementations. Ok to check my understanding: Sigma-Delta DACs employ zero-order-hold OS(repeating samples, correct?) while the Denafrips is adding samples equidistant between those input? This would mean both methods have new samples however in the zoh method the samples are clones of those input while they're novel with linear interpolation? As a follow-up question; are the polynomial filters in HQP similar in theory as to the linear interpolation in the Denafrips? I've been playing around with the trial of this software trying to find something comparable to the major sound quality improvement I hear running the X16 in it's NOS mode and reading the included documentation these sounded similar in theory: not NOS but following the NOS promise of preserving the impulse response.

I've been following along with this discussion as a NOS diehard whose been playing with other gear as of late & thinking of making a return. I'm not really sure what to think of the whole situation, it seems on the face of it the Ares is being advertised as something it's not BUT we also have a long standing precedent now of Sigma-Delta DACs which include a NOS mode which can be used with standard Redbook content. How are these functioning? Are they likewise Linear Interpolation? If this is the case then it strikes me that Denafrips isn't doing any unusual and have simply labelled this mode by established norms. I wonder because I recently ventured into experimenting with the NOS mode on my Gustard X16 DAC(Sabre ES9068AS chipset) and find that it achieves the effortless sound of NOS once the treble is properly restored with EQ(36db(!) down @ 20khz.) It likewise avoids the audible aliasing of your standard unfiltered NOS DAC(tested subjectively with frequency sweeps.) I'm quite curious what my DAC is doing and if the Ares or other DACs* might be doing something similar as what I'm hearing right now seems a clear 1up over the traditional way of doing things. *I know AKM has something similar with their super-slow filters, to my ears those don't sound very good. They don't remind me of NOS R2R, that's why I'm shocked what I'm hearing from this Gustard.

You upgraded how recently? I'd listen to the system as is for awhile. Once you can pin down the weaknesses of your system and where you'd like to see improvement it will be a lot less a shot in the dark when you choose your new DAC.

I'm limited in filter selection to the 'poly-sinc' and 'minringFIR' filters. I'd be interested in hearing what filters people prefer for their own use, curious to know where the filters I have use of line up quality wise with the others. As my DAC has no digital filtering I'm safe in assuming there is a discrete analog filtering being done in the signal path which certainly would not be defeatable? Impressions so far: All of the filters add a subtle fullness/wetness to the sound. Added distortion or removed distortion? Linear filters seem to add more however I've not found this always obvious in listening tests. Minimum phase filters have a more coherent transient however instruments can become incoherent in the stage leading to glare. Preference for filter relies on genre of music, I've been switching between them unable to determine a clearly superior choice.

To followup on my first post(my being the proponent of filterless NOS DACs) I'm now evaluating HQplayer as was mentioned earlier in this thread. As my DAC won't accept anything other than 44.1/16 I am limited to the poly-sinc filters. Or perhaps it is another reason, I'm not sure. To keep it short, I do not notice any obvious shortcomings using these filters. Vocal legibility(the simplest thing to test for) remains consistently as good as going direct. The sound is certainly different, the brashness/grit of my UF DAC drastically reduced. As such the sound is different to the point further evaluations will take time. As I do all of my listening with Jplay I'll need to control for this variable as well, by testing HQplayer filterless and running HQ through Virtual-Cable into Jplay as I do when streaming online. How does this approach work when using DACs with filters built-in? Such as the Questyle I had before, surely the DAC would employ another layer of filtering over the software? I didn't see anyway to disable it.

Filtering may be the most important factor to consider once you reach a certain level of hardware. Currently I'm a proponent of filterless DACs, these DACs happen to be(marketed as) NOS however after testing a DAC with defeatable upsampling it's not something I'd focus on for a purchasing decision. Because my UltraFi DAC is filterless I have "ringing [which] adds additional energy, seemingly making the music louder, bringing the sound stage closer and making the overall sound livelier." Undeniable, I've described my UF by those words before. That is the downside to this approach. I had the fortune last year of having a Questyle CAS192D for a few months and I spent much of my time home swapping between the two DACs and the various filters on the Questyle. The sound of the Questyle vs UF would fits the stereotypes attributed to tubes & solid-state. The UF is brash, fast, loud, and a little gritty; the Questyle is rounder, fuller, fluid, and a touch glossy/wet. If the UF showcases compression; the Q showed normalization. Separation, size, weight, flow, and imaging were superior on the Questyle; However micro-detail, timbre, vocal legibility, transients, and elasticity were lost in the filtered DAC. I found each filter on the Questyle compromised in one way or another. All of my complaints regard timing and gloss. The Questyle reminded me of my cassette deck; IIR filters suffered 'flutter' and FIR filters 'wow.' Jitter vs syrup? This were obvious on the kind of test I do for timing, I listen to recordings of campfires. It sounds absurd however I find the pop/crackle of a fire an ingenious way to test speed, Its not a musical event where room acoustics come into play. We should all have a similar reference of what a fire sounds like. The UF adds unnatural contrast however it always keeps perfect speed. My conclusion, the filters built into the WM8741 DAC chip are insufficient. The loss in grit/legibility to vocals & slight instability in speed taints an overall refined/cohesive/well-imaged sound. Given enough time I suspect newer chips will overcome the performance ceiling of the ringing unfiltered approach. They may have already, The Wolfson DAC I tested isn't the newest tech. The UF's DAC chip remains a mystery. I'll wait for a few generations of DAC chips to pass before giving it another go. I use exclusively 16/44.1K tracks; perhaps things are different with superior files/formats.

I'm young but my stereo went through a phase where all my components were vintage(not counting DAC) so that'll have to do. STAX SR-5 Headphones Sherwood S6000 Reciever Tandberg TCD300 Cassette Deck Dual 1209/701 Tables

I received an email back from them and was quoted $1359 on the 16-bit DAC. The volume control is $860, awaiting more information on the yet to be released amplifier. There is currently no option for SD card reader however they are working on an external unit which outputs through toslink. From a quick search around the internet I picked up on that they sell direct and are overburdened with orders at the moment.

They've good timing, with all the devices on market designed to improve the quality of the USB connection it's a breath of fresh air to have someone pop up and say "forget about all those little boxes, we've something better that stands on its own." I'm talking with them now and while the technical details go over my head I know this will be an easier solution than upgrading my DAC and all that lies between it and my PC as I won't be concerned about mismatches and extra cabling.

A regen might give my DAC a new lease on life, save up some cash for a new amplifier. Because let's be honest it'll end up being spent on audio either way.

Intriguing, I'm getting fatigue with track C whereas neither A nor B had that effect on me. I cut my testing short this time and will stick with the perception that D is a pleasure to listen to and the notes more clearly defined.

I got this working on the main rig this go round. A. The echos on the chimes sound more floaty to my ears and I hear less overtones with the drums. By that I mean on B the bass is more 'bass note' than a distinct instrument.

That's a whole new level in minimalist marketing. At least when the product is in the photo there is some reassurance they didn't just lift a stock photo. So the consensus is that the distortion is probably limited to the low-end chips and even then it doesn't come up? I'm mostly concerned as my collection is not mastered well. The loudness war hit the genres I like hard; if this flaw is real I'd be running into it more than not. And being that reviewers use music that pulls the most from a test system I can see many of them never experiencing this problem.

I was reading a review of the Henry Audio USB DAC.. I'll let the quote speak for itself "In your first iterations of the DAC, you employed an ESS Sabre chip. The new DAC has gone with a different chip. Why? "ESS don't make their products available through typical electronic parts distributors. Getting them in low volumes was a big hassle. But what made me choose another chip was the discovery of an internal math bug in the ES9022/9023. A >-1.3dB full-scale square wave played through their chip will generate great amounts of distortion. With modern compressed music, that isn't just a theoretical occurrence. I worked many years with signal conditioning and was able to see what goes on. In technical terms, the internal FIR filter does a 2's complement overflow where a very positive number actually flips around and becomes a very negative number. This occurs before the sigma-delta modulator. A digital limiter or lower gain in the FIR would solve this." - 6moons audioreviews: Henry Audio USB DAC 128 MkII I'm currently in the market for a new DAC, and stumbling across this brought my search to a halt. Much of my music collection is sadly highly compressed. If I'm going to remove all Sabre DACs from consideration I don't want to do so based on a single account. I was wondering if anyone else has tested this theory or has the background to look over the whitepapers and determine if there is any truth to this. Most importantly if there is anything to this; is it an issue in all Sabre chips or specific products? I have experience with a DAC using the ESS 9010-2M and whilst I did find the sound of it far from neutral I feel that I'd be a bit like an inquisitor trying to make the problem fit the impression.