Audiophile USB to PCM

I decided to go with the Waveterminal U24. My understanding is that it is very well made and will provide a bit accurate stream with no conversion from 44.1 to 48k. I'll report here what I find.

Thanks to all for your kind help.

harry

Here's the Edirol piece to which Ed is referring. This is the reverse of the piece I posted before, which means it is what I thought the other piece was. It will take a USB connection from your computer and convert it to SPDiF or Toslink, keeping the signal in digital. But, nowhere does it say it converts the signal to PCM. I'm not sure where this is coming from. My understanding is that the computer puts the signal out in digital at whatever resolution you imported it and I don't know that it needs to be "converted" to PCM nor do I see that the Edirol piece performs this function. I'm not sure what the problem with the M-Audio piece wasm either -- but -- I can tell you that when I was running MP3, the pieces in my system were not critical. I could take a feed from the headphone out, split the signal into L & R, feed it into analogue inputs in my pre-amp and you couldn't tell much difference. Now that I import my CD's uncompressed, you can tell large differences between, for example, the M-Audio piece and the Apogee.

http://www.edirol.com/products/info/ua1d.html

The Waveterminal piece is interesting. It offers sample rate conversion. Maybe this is what is needed to take the signal from the computer to a home audio DAC and maybe that is what was missing from the M-Audio piece. Like Harry, I am no computer expert, I learn as much as I can in order to get the computer to do what I need, but my interest lies far more in music, which is the carrot that leads me to learn about computers. I am curious to find a product that will let me connect the computer to the digital input on my pre-amp so I can use the DAC in the Proceed, so I will be interested in hearing about Harry's experience with the Waveterminal piece. I the meantime, I am getting a real kick out of this G4, Apogee, Mackie system I've built. It is surprisingly accurate, musical, and enjoyable. But, this has opened up a little mystery and now I am curious to know whether uncompressed music files like AIFF need to be converted before they can be read by a home audio DAC and why my Apogee DAC doesn't need any conversion before it can perform its magic.

Rsbeck:

Thanks!! Do you think any or these converters are better or worse than each other?? There are lots of them; M-Audio, Edirol, Xitel, Apogee, Ego Systems, etc.

Also, I'm interested in the differences between streaming through a network as the SlimDevices Squeezebox does, and the hardwired devices listed above. I guess the real issue is what is going to provide the most bit accurate stream most effectively.

harry

Going back to basics, there is a bitstream on the CD and the idea is to feed that to a DAC to convert it to analog. PCM just (I believe) means pulse code modulation, which basically means communicating bits between two devices by an agreed upon standard for interpreting voltage pulses on the wire between them. USB is a serial interface, which means among other things (I think) that its buffered, so its a bit more hardware to implement than something translating pulses into TTL signals for digital processing. So, because coax digital is PCM, and USB is specific serial protocol, there is a conversion from a serial protocol to a different electrical format. Hopefully doesn't mean that the actual datastream of 1's and 0's is changing, but when I've got a box that is USB on one side and coax digital on the other, I call that a "converter."

Of course, I could be dead wrong. Its happened before. ; )

AIFF/WAV/MP3 is just a means for storing the bitstream on media. There is all sorts of stuff added to store things--directory records, sector sequence data, etc--even if you aren't doing any compression. But, with compression, you also need software to recover the original bitstream (or something that looks pretty close, if it was lossy compression).

I wonder, based on what you said, whether your M-Audio somehow ran the bitstream through a low end DAC and, if you used the digital out, ran it through an analog to digital converter. Doesn't seem to make much sense to me, but might explain the problems.

Sample rate conversion sounds sketchy to me in terms of rewards... Can't get more samples per second out than there are samples per second in. But, that is what oversampling is and that seems well beyond my technical competence. How does the thing *sound*? I guess the benefits of creating a 24/96 stream out of a 16/44 stream is going to depend heavily on the quality of what is doing the conversion...