I have finally finished archiving my entire CD collection (2500 CDs). I converted all over to FLAC format using EAC. I use foobar and winamp for playback. I can't stand winamp and prefer Foobar. One thing I notice is that I can consistently find differences between straight CD playback and playback of the FLAC files. I'm running Digital out (COAX) from the PC to my preamp/processor (Bryston SP2). I suspect that the problems are coming form the sound card on the motherboard. I am looking to purchase another sound card. I'm after something with an optical out and also support for DVD audio, DTS-HD and Dolby True HD. I contacted some vendors (Creative and Auzentech) but their answers are completelty useless. Can anyone make a recommendation?

Don't open multiple topics with same question!

Possible causes:
-sample rate conversion 44.1 -> 48kHz. Find a card that can do 44.1 out on spdif
-not bit-perfect digital out. I prefer not being able to control volume, using things like kernel streaming i/o mode. When your equipment handles DTS wav files, trying them is easiest way todetermine if setup is bit-perfect.
-spdif jitter.

MLP decompression is more on the software side. The best way would be to wait for usable free software to appear. But you will never be able to transmit full resolution multichannel through Toslink. Downmixing to stereo or encoding to lossy AC3/DTS must be performed.

Given the opportunity I want to say how satisfied I am with EMU 0404 for stereo analogue I/O and bitperfect S/PDIF (even on WinXP).

read this thread completely

http://www.hydrogenaudio.org/forums/index....showtopic=59825

DualIP

Thank you the problem is 48K out the SPDIF port. The resampling to 48K cuases some horrible aliasing issues. Forgive me if if I ask dumb questions, I am new to PC audio, I always dismissed it due to the crap hardware and lossy compression. What got me into it was FLAC and the availability of large storage devices. Why the hell would anyone pick a sampling rate of 48K (let me guess - it was MS)? I am using the motherboard sound card but it seems to me that it would be far easier to pass through the digital PCM stream at 44.1 K rather than resampling! Is there anyway of setting the sampling rate/clock rate on an Asus MB? You mentioned handling DTS wav files - how can I determine from this if the sound card is bit perfect? I am looking at the Auzentech 7.1 Prelude. Build quality looks good ( i work as a manufacturing enginneer for microwave electronics - mostly cellular) and I can upgrade the op amps. I am after a rock solid clock with pass through when required but also the ability to do onboard decoding of multichannel DVD-A and other content. Soundcards are cheap in comparison to the rest of my system, buut when I look everything seems geared towards gaming or home recording, neither of which I am interested in. As a sidenote the Bluray player in the PS3 is kickass. There are BD players costing much more that cant touch it and one can upgrade from 1.0 to 1.1 to 2.0 with FW updates. This can't be done with any player that I know of.

This is not the place to bash Microsoft. Audio DVDs and movies (AC-3/E-AC-3/DTS/DTS-HD) also usually have sampling rate of n*48 kHz. 44100 and 44050 are actually stupid numbers derived from VHS video tape, when used as audio storage medium. Problem with cheap audio interfaces is that their clock rates can't be adjusted. More expensive gear can switch between 44 and 48 easily.

Decrypt your DVD-Audio discs with the newly released DVD-A Explorer, flac them up, and listen to them in Foobar.

Unbelievably enough, its the most hassle-free way of getting them working.