ConcernedListener

Thank you for the responses, gentlemen. You're right, I should have elaborated more about what my goals are. I'm a huge fan of vintage hi-fi, but it appears that the wave of the future is coming in the form of high-resolution 24bit recordings offered by HDTracks, Linn, 2L, etc. All I wanted to do was figure out a way to connect my hi-fi to my computer to listen to this music. I decided to test my feet in these waters first without breaking the bank, so I bought an HRT Music Streamer II+ and some quality cables to connect my computer to my amplifier. At first everything sounded great, but as I said in my original post, I noticed slight variations in what I was listening to depending upon what player/plug-in I was using. As a musician, its driving my ears insane. I just want to hear the music the way it was meant to be heard - hence my want for bit-perfection. Also - there is no specific reason why I want to / am using JACK for playback. I'm still trying to find something better. I actually just saw the article about Voyage MPD the other day on Computer Audiophile and became intrigued. I've read that MPD can be set up with a client on the same computer (GNOME client or Sonata, perhaps). I might try that before I start building a headless machine. And Miska, I understand that playing 16-bit files through a 24bit/96khz DAC would not be bit-perfect, which is why majority of my music is 24/96. My question, better worded, would be that if the bit and sample rate of the file matches that of my hardware, would JACK output that as bit-perfect? I don't see why it wouldn't, unless some type of volume control was being used on the computer (in other words, not at unity gain).

In that case, perhaps my direct to hardware, bit-perfect answer does exist in JACK after all. In the JACK setup I can select a driver (ALSA) plus an interface (hw:1, my USB DAC). Are my assumptions correct? If not, any advice on how to achieve bit-perfection on Linux / players with "smarter" ALSA plug-ins (no mixers!)?

Thanks! Keep 'em coming!

Thank you, Miska, for clearing that up for me. But now it leads me to a new problem! I would like to achieve bit-perfect playback, and I know it can be done with Linux, but the problem I run into is that all ALSA plug-ins are looking for a mixer in order to run. If one isn't specified, obviously the audio won't play. Do I need to bypass the DMixer, perhaps? Audacious does have the option to 'check-box' the setting "Bypass all signal processing if possible" - which sounds hopeful at best. If you don't mind, could you expound on the idea of direct interfacing with ALSA?

Here's the situation: From the information that I have gathered, I assumed JACK audio was bit-perfect. Recently, however, I have been playing with different media players to decide which one I like best, each with their own JACK plug-in. I have noticed subtle differences in the sound that is produced through different players. For instance, through Audacious w/ JACK plug-in, the sound is somewhat "brighter." Comparatively, playing the same song through players like MPlayer or VLC (also w/ JACK plug-in), the sound is more laid back, and the bass is more pronounced. My questions: 1. Is JACK actually bit-perfect, or have I been deceived? 2. Are all JACK plug-ins created equal? Each player has their own, but are they all essentially the same? 3. Because I have to set the sample rate under JACK before I start the server, does that mean that files played below that frequency are re-sampled? (e.g. My USB DAC can handle up to 24bit/96khz files, so naturally I set JACK to 96000. If I play a 16bit/44.1khz file, does it get up-sampled to 96khz, or does JACK play it as is?) If we can come to a conclusion on this matter, I may be able to sleep easier at night. -ConcernedListener