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I've been out of the loop on Linux audio for a while, but with a new NAD M51 I'm interested in trying out USB. I have an old single-core Atom Acer netbook that would be used for this. I know Voyage is popular for dedicated audio setups, but a) I don't currently have a linux box to help with the install b) Doesn't really have a live option for quick testing from what I remember. Is UAC2.0 firmly enough entrenched a this point that I can just grab the latest Ubuntu or something and have reasonable confidence that it will work, or is there a particular flavor of general distro that might be a better choice?

I've had a chance to do a bit of listening, and my initial impressions are that the HDMI input doesn't seem to be up to the quality of the optical input. Redbook over optical from an Oppo 980 sound really good, but SACD->88.2k PCM over HDMI doesn't seem to be as good. I'm not all that familiar with the SACD content though, so it's a tentative conclusion. I hope to try some spdif vs usb tests soon-ish, but that'll have to wait until I get my squeezebox back in action.

I recently picked up an M51, largely based on feedback here along with a real interest in the technology. This is my first top-flight DAC, so I'm not sure I'll be able to offer up much in the way of informative comparisons. At first blush it seems to be a clear step up from my Bifrost, but then again I would hope so given the price differential. On the plus side, it sounds great running directly into my new Stax 3050 setup. On the minus side, it turns out you do need a monitor attached in order to get hi-rez over HDMI. Without a monitor, my Oppo 981 only puts out 44.1 when playing a SACD. With the monitor for handshaking, you get 88.2 as expected. I'm not sure how I'm going to use this yet. We're in the process of trying to get our house ready to sell. There is a chance I may use this as a "2.0 pre/pro" to handle both audio and video in a pure stereo setup. It may be a more normal audio-only unit too, depending on how things play out and where we want the 'good music' system - living room or family room.

"If you go for the multiple DACs, I would definitely run them off a single clock. " Why are people recommending a "single clock"?? In this case he's suggesting running them all off spdif feeds from a single sound card, which by definition will provide a single clock for all 3. I tend to agree that I'd lean towards something like the Lynx aurora or the LIO8 rather than an AES16 and 3x stereo DACs, but what he's suggesting will work fine.

"...Like what? I am looking at this for similar DSP / DRC reasons" I meant 'simpler than dsp active 3-ways', not 'simpler than the e18' if that wasn't clear. I do think the e18 is on balance the most attractive option for a PC based DSP setup right now when coupled with JRiver. If this was for a dedicated audio setup or even a dedicated 'theater' space, I'd probably be leaning that way. As I said in the 'DAC' thread though, this is for a 'main living area' setup which means a) I'm not sure the PC-only paradigm works since TV/sports is the majority of our viewing over movies b) Has to be usable by the wife c) I want something more 'done' and less 'project'. I want to spend my available time listening, not building. Point c is a big one at the moment - I really am looking to take a step back from DIY for a while, at least to the point of not having my 'main system' being a constant project.

"2) Am I correct that the s/pdif handles only 2-ch? Why?" I'm not George, but have corresponded a bit on the e18. SPDIF is 2 channel only. The reason(s) would be - I expect he's using the built-in Sabre implementation for handling spdif - Handling multi-channel HT formats requires huge $$$$$ in licensing costs. I think you have to understand that at it's core, this is an enhanced exaU2I interface with a Sabre DAC grafted on to it. The entire premise of the exaU2I was to be an audiophile grade multi-channel async USB solution, and external sources weren't a consideration. So, the e18 seems to be pretty much targeted at the same audience - one for which the PC is the only source. JRMC is capable of doing blu-ray audio decoding with a few tricks, so the e18 + JRMC combo should suffice for top-end playback, and should do better on the audio front than all but the top-end mega-buck pre/pros. Of course, this doesn't cover cable/satellite sources, xbox etc. I find the e18 very very intriguing for a DSP based xover setup. Having been down that path once already though, I think I'm headed for a simpler solution.

"There are significant feature differences between the two (e.g., DSD and FireWire on the Mytek vs. HDMI and the whole "direct digital" thing on the NAD)" True - and taking it a step farther the QB9 is usb/computer only with no external source capability etc. System architecture and interface will definitely factor in. In my case, it's definitely a big factor, maybe the main one. I'm looking at a living room/family room system rather than a dedicated audiophile setup, and the NAD fits that much better. In fact, I'm really intrigued about it being effectively a replacement for a pre/pro as long as you're happy with 2.0 in your video system. What better way to finally get audiophile performance out of your video system - just go stereo :-)

"3) Everyone thinks they have the best DAC and made the best buying decision at the time." Well, yeah :-). Kinda follows from my point a) but I was trying to be diplomatic. Hopefully the good news is that it seems like none of these is a bad choice.

Aren't we seeing this with all of them, though? Not that I question the sincerity of the comments, but isn't it pretty much standard audiophile discussion fare that components are said to compete with 'much more expensive units' etc etc? IMHO this reflects one or both of the following a) the units are actually very close in performance, and we struggle to create meaningful differences that aren't really there, since that's what audiophiles do b) they're close in absolute performance, but it's a matter of 'flavor'. All of this points to the importance of audition to determine which you personally respond to, but that's becoming increasingly difficult. I'm strongly leaning towards the NAD, but that doesn't mean the Mytek isn't making a case for itself.

If you can live with the PC as the only source, the e18 from exasound merits consideration for an active xover setup. The exa guys have the async USB interface pretty well implemented, and assuming their Sabre implementation is at least competitive with others I think it's an interesting option. Built-in volume control, and consumer friendly output format and levels rather than the pro levels you get with the studio DACs. The big downside is that it's a new product from a young company.

I'm looking in the $2k range rather than 2.5k, but the NAD M51 is the one catching my eye. Not a ton of feedback yet as it's new, but positive so far and I find the technology intriguing.

Sure. Most spdif tx/rx chipsets work with I2S. I'm fairly certain that the Twisted Pear spdif receiver based on the WM8804 can be configured as a transmitter, fed from the I2S on the exaU2I. Gets a bit pricey, but it should work (as long as the part is in stock at Twisted Pear). You could do it cheaper using just the raw chips if you could roll your own boards and/or do it on perfboard, but I'd be a bit concerned about compromising performance with poor layout etc.

I recently went through a similar decision process, looking for an interface to use as part of a multi-way DSP crossover system. I was originally looking at the Lynx Aurora as the top-end of my search, but after an honest assessment of my budget I ended up with the Steinberg MR816x. It's limited to 96kHz, and so doesn't satisfy the 192 requirement, but otherwise based on the overall chatter from the studio-oriented forums it seems to deliver a significant fraction of the higher end converters for substantially less money (although I don't know how the relative pricing plays out in Europe). I've never directly compared it to anything in my current system (multi-amped DSP systems being rather tricky to A/B), but am very happy with the decision. Lowering the gain on my amps allows me to run a direct balanced connection from the MR816 to the amps, relying on the MR816 to handle volume control and eliminating the need for an additional volume control stage in the mix. The MR816 also has 2 independent spdif outputs and so in theory could enable future expansion via external DACs as funds allow. Another option to consider, anyway.

A brief description of my system. I ended up in a rather different place than I expected when I started out. I've been tinkering around with FIR filters for ages - going back at least 10 years with BruteFIR and Linux. I've never had a 'production system' though. I spent a lot of time trying to get a Linux/BruteFIR system running with acceptable quality and ease of use. The closest I came was with my Yorkville U15's and the Emu1820m, using BruteFIR only for DRC/room correction filters. This actually sounded fantastic, but then the PC died and took all my filters with it, followed shortly after that by losing my dedicated listening room to office requirements. My new plan became a nearfield/desktop system since I work from home a day or two per week which represents valuable potential listening time. This really drove my 'listening requirements' though, since I use my company laptop hooked up to my monitor - this makes using dedicated GUI based programs like Foobar a problem since I don't really want to switch back and forth between computers. Remote solutions like MPD were an option, but I soured on Linux for audio when my 'fully supported' Saffire Pro 24 absolutely refused to work in any setup I tried. I have been using Squeezeslave on the PC since it supports ASIO and is headless, but it only supports 16/44 and so isn't a viable solution for hi-res. The other constraint is that I got an Atlona HDMI de-embedder to enable conversion of SACD to 88.2kHz pcm. My intent was to 'record' this onto the server rather than play it real-time, but it then occurred to me that being able to feed in arbitrary content in real time from a DVD player (maybe blu-ray down the road) was a much simpler prospect than figuring out how to get VLC or something cleanly integrated with the system. (we use this system for mind-candy content while working out, hence the video requirements) So, I actually ended up using external sources rather than a 'pure' PC setup. Music goes through a Squeezebox (currently an SB3, but I'll get a Touch), video/SACD/DVD-A through an Oppo 981HD and into the Atlona de-embedder. Both output digitally. External sources bring up the 'sample rate problem'. I solved that with a Behringer SRC2496 hardware sample-rate converter. This is slaved to the wordclock output from the MR816, so the audio interface remains the master. I've tentatively chosen 88.2k as my global rate since redbook and converted SACD will be my dominant sources. The audio interface is the Steinberg MR816X as mentioned previously. Amps are the Class-D-Audio SDS-224 for the mids and tweets, and an Alesis RA-150 for the woofers. Speakers are a pair of the NHD Xds monitors from the NHT close-out liquidation sale. These are paired with a pair of Dayton RSS-210 woofers. For Software, I'm using Reaper (www.reaper.fm) which is a very nice DAW setup. I haven't yet gotten the FIR filters set up, but will be using Pristine Space for the convolution. I've currently dialed in some IIR filters using a parametric EQ built in to Reaper - the filters are based on the crossover the John Krute (zaphaudio.com) put together for the Xds monitors. Reaper has one killer feature which is the ReaRoute virtual ASIO driver - this enables directly capturing the output of an audio program and routing it into Reaper directly as an input channel; this leaves open the possibility of moving to or adding software-based players which would bypass the external hardware SRC if I ever start to feel that it's a limiting factor. So, this is certainly a different style of system than some of the purist systems discussed here, but initial listening even with xover filters that haven't been fully optimized is very very promising. Certainly it's by far the best system I've had set up since the optimized U15 system. It definitely got more complicated and expensive than I originally envisioned, but on the other hand I actually am feeling that the decision to go with external sources via the SRC2496 was a good one - it makes things more of a dedicated system rather than an piece of my office PC, which I think will make for more focused listening rather than 'listen while wasting time browsing the web'. I'm hoping to have the first pass of FIR filters done this weekend - there are tons of measurements of the system around, so for the first pass I'm simply going to clone the transfer function of the original DSP unit used by NHT; they basically used brick-wall filters @2.3k, but incorporated some baffle-step compensation and a bit of driver eq. After that I'll get HOLMImpulse set up for some more refinement

Hi Aps, Well, to start my comment on video was very straightforward - I was thinking that dedicated audio players don't support video. I neglected the fact that JRiver handles both. So, from that perspective making a commitment to JRiver and embedding the VST may work well. Certainly, the complexity grows rapidly if multiple playback applications are involved. You're correct that the latency from FIR filters may be an issue, both due to the processing latency of the convolution plugin, but also due to the filter itself - for linear phase filters the impulse peak is in the middle of the filter which imposes an inherent latency. JRiver may have latency compensation of some sort though - if it can delay the video sufficiently then that would be the ideal solution. If not, then there are a number of possible ways to address it, but none are entirely ideal. I believe that Pristine Space uses a variable partition size scheme to reduce latency which might help, but nothing can compensate for the inherent delay present in linear phase filters. ConvolverVST supposedly will auto-switch filters based on sample rate. IF this actually works, then it may be possible to have filters set up for the 48Khz video rate that are not linear phase and hence have lower inherent latency. I believe there is a config example for this either on the Convolver site or maybe on the Acourate Yahoo Group page (this is a resource that may be worth seeking out in any case. A lot of the Acourate guys are using RME cards under Linux with BruteFIR, but there are some PC/Win guys there as well) Finally, you could use standard IIR filters when playing video. There are a variety of good parametric EQ VST plugins that can do conventional crossover transfer functions, and will have minimal latency. Once again though, it would entail switching configurations which is a pain, and doesn't play well with 'other family members'. The MSB MVC is a nice unit, and works as advertised. I admit that I never did dedicated A/B comparisons, but informally the quality seems very good. Having said that, I'm not actually using it in my current set up, since I'm tentatively using the DSP volume built into my interface - that may not pan out in the long run though, so the MVC is on stand-by notice. The main question I'd have on the MVC is whether a single unit has enough channels for your application - if you want 3 way mains, then 8 channels may not be enough. I believe that multiple units can be linked/slaved though, although it starts to look like less of a bargain if you need two units. For me, the benefit of using DSP volume control is being able to use a direct balanced connection from the interface to the amp. The amps I'm using (Class D Audio SDS-224 kits) can be made to have sufficiently low gain that attenuation isn't excessive, and they have a good balanced input stage based on the THAT chip. My previous experience with computer interfaces (with an Emu-1820m) was that they really seem to perform better balanced than single-ended. Maybe all in my head, but it doesn't seem unreasonable that PS noise etc may present itself as common-mode and hence less of a problem with balanced connections. Your list of possible interfaces is certainly a good one - the Lynx Aurora is the one I was originally aspiring to, but after some soul-searching I had to admit that the price tag wasn't something I could swing in the foreseeable future. I ended up with the Steinberg MR816 which seems to be the consensus pick for audio quality at the mid/low price point - at about 1/3 the price of the Aurora FW it was an easier reach, and it really seems that the opinion in pro circles is that you have to move up to the Aurora or the Metric Halo units to noticeably improve on the converter quality. Initial impressions are very positive. The idea of adding a dedicated DAC for the midrange is there as an upgrade path, but I'm hoping that won't be necessary.