WAV vs. FLAC vs. AIFF

Mapman - whether you can hear these differences or not is highly dependent on your system. For instance, if you use an active preamp of any kind, you may not hear a difference. Preamps add a layer of noise, distortion and compression that is significant. Until you run without a preamp (and I dont mean with a resistive passive linestage), you will not realize how much grunge your pre actually adds.

I always believed that my highly modified Mark Levinson Pre was really transparent. Then I built a DAC with a good volume. Boy was I wrong. The pre is now gathering dust...

Steve N.
Empirical Audio

I use network players designed to stream audio as the source feed to the dac. That keeps any issues that might be associated with using a general purpose computer as the source out of the picture.

With this approach there is no audible difference.

OTherwise, there are many factors that can come into play that affects sound with any source type for that matter. Power/jitter issues associated with decompression processing can stand in line with all the rest.

But the format itself does not correlate to sound quality in general though. Lots of other crap can go wrong and chances are it does so differently because of different hardware and software processing scenarios for different formats. The devil is all in the details. But not in the source format itself. If processed properly, teh results are the same. That can be a big if though.

Personally, I prefer .wav. Probably lower risk in general but not inherently better or worse otherwise.

Roku, Logitech, .wav, flac. It all sounds essentially the same and quite excellent to the point where if there is a difference it is not an issue at least for me.

FWIW, I can change most anything else in my system and hear a clear difference, including ICs, but none at all with any combo of Roku, Logitech, .wav, FLAC.

It also doesn't matter what kind of computer I use for the server. I've used various notebooks over teh past few years. They all sound the same witht eh network player approach. The only issue is if they have enough memory and CPU speed to stream in real time without rebuffering at the network player occurring and how fast library scans and such take. Squeezeserver on my current 8 Gb Gateway laptop can completely reload its music library from a USB disk drive in about 10 minutes (1700 albums, 18000 tracks, 99% .wav, 1% flac and mp3 downloads so far).

Roku Soundbridge is an older and curently poorly supported platform, so I do not recommend that these days, but otherwise the sound quality through a good DAC is top notch as is Squeezebox Touch through the same DAC (I've used several....DACs make a HUGE sound difference, so worry about that first).

Mapman - that figures. With networked audio, you are less likely to hear differences. Data is just data, not audio data as with USB or Firewire, so the audio stqck is less involved, if at all. Jitter from the computer is also a non-issue with network playback. Only the end-point device jitter and clock matters.

Steve N.
Empirical Audio

AE,

Exactly!

The network player essentially serves as a "proxy" and effectively isolates the DAC and the rest of the audio system from direct interaction with the computer.

ITs definitely the way to go for those who would fret about how well their computer might play in a high end audio system, assuming your network has sufficient bandwidth.

Most home wireless G networks with moderate to strong connections should work fine under normal circumstances unless others in the house are in heavy competition for bandwidth. Squeezebox Server (now Logitech Media Server) actually converts files to lossless but compressed FLAC to make better use of network bandwidth and this solution works very well. Roku soundbridge does not but also tends to rebuffer more frequently with weaker connections in that data is not compressed before transmission. Roku Soundbridge and Squeezebox Touch both sound essentially identical in my rig.

Steve - It seems like what you are describing is a timing problem that comes from the competing flac decoding and audio stack processes. How does that timing issue get passed to the DAC. As I understand it, the async USB is suppose to control the timing of the data received by the USB async converter using its internal clock and it takes the computer timing out of the process. If their is a timing error/latency from the audio stack and flac converter competing for run time that is coming through the aysnc USB converter, then it seems like the async USB converter is not effectively controlling the timing of the data. Are you suggesting that the run time issues make the aysnc USB process not work properly? If you think the aysnc USB process is working properly, then why would the flac and audio stack processes introduce timing error in the aysnc USB device?