Hi all,
I'm a total beginner with the more involved side of hi-fi, so please be gentle...
Firstly a big thanks to Spanky and everyone else for the detailed discussion on the Maverick...I've recently decided to splash out on a dedicated DAC - basically, I bought the new Peter Gabriel album on CD, listened to it and wasn't overwhelmed...but it came with a code for the 24-bit recording and it really opened up (even did a blind test, though my housemates probably think I'm mad now)...so I kind of figured that it can make that much difference on a system with basic functionality, I might see another leap forward if I invest.
I was tossing up between the V-DAC, the AUNE, and the Maverick...obviously comparing apples and pears as they're all pretty different in terms of functionality etc, but I'd love to have a bit of tube equipment (love the analogue sound), and I'm essentially going to use this for PC media, and then possibly a CD player or SACD if I upgrade later. So I'm pretty set on the Maverick (although I live in the UK, hopefully won't get stung too badly on customs, I will email Ryan and ask of his experience with UK customers)
My main learning curve has been on the bit-perfect/matching concept. I have onboard sound with a Realtek 885; seems this might actually be quite useful, as you can select the samplerate, although it's fixed once you set it, so to get bit-perfect you have to change it if you're listening to a 44.1 then a 48 in the same sitting. I've also installed foobar and ASIO4ALL , and have kernal streaming via a program called ReClock. If/when I get the Maverick, I'd be connecting the PC via a Coaxial S/PDIF.
a) is selecting kernal streaming via ReClock redundant if using ASIO4ALL? Is one method better than other?
b) Can I confirm I have my thinking right: In my case, where the Realtek manager sets the Sample rate, ASIO4All won't bypass that and just output at the file's native sampling rate; it will just bypass the windows mixer and resampling. so the signal would go Foobar-> Realtek control -> out the SPDIF
c) I had a bit of an idea - could you connect both the USB and the SPDIF of the PC to the Mav, and select the SPDIF on the unit when playing 24/96 files, and the USB when 16/44.1? As the USB is recognised as a device seperately from the soundcard, so seperate from the S/PDIF...but then I suppose in Foobar, for the 24/96 files, you're forcing 24-bit depth, so you'd be sending 24/44.1 out to the DAC. But that might be preferable to resampling 44.1->96 before the DAC at least? Or you could have a seperate media player for your 16/44.1 I suppose. I guess what I'm getting at, is that I expect it's prefferable to have the minimum amount of adulteration to the signal before it gets to the Maverick - so the least amount of resampling - but also without the slight hastle of having to click a load of different settings to get bit-perfect for a range of different files (I am probably just wanting far too much for the price...but I am going to buy it anyway, just trying to consider how to get the most out of it!)
d) Just to check aswel, a bit of hardware or algorithm set at say 24/96 won't put a signal already at 24/96 through a process anyway? Probably a really silly question, just wondering in terms of my Realtek control forcing a sample rate, making sure that it will just let a signal already at that sample rate pass through (a basic analogy would be say take a number , +4 then -4...except if the process isn't actually perfect, it might be like +4.000001 then - 4.000002...if you catch what I mean...).
Cheers for any help, I appreciate it's a horrendously long post, just wondering on thoughts if anyone has the time!