Hi All, I've been struggling for a while now trying to figure out how to send sample sync information from an existing soundcard in a PC to a custom digital audio input device (a multi-channel digital guitar project). The idea is to be able to run a PLL in the digital guitar so that its ADCs are sample synced to the regular PC soundcard, without resorting to solutions such as a HW word clock signal. I don't need a high degree of phase accuracy between the two audio systems or anything like that. I just need to keep the ADCs from drifting apart over time. The problem is that the Windows OS doesn't support this sort of concept. (Apparently the Mac doesn't either, but they have some built-in sample rate converter code to make things workable). Anyway, I had this nutty idea of solving the problem by building a piece of interposer software that stacks on top of the existing soundcard driver, and taps off a "buffer clock" signal (sent whenever the soundcard needs to source or sync new audio data). The buffer clock signal would be sent to the digital guitar as an input to the PLL, and the interposer SW would merge in audio data received from the digital guitar. So the interposer SW would appear to the original soundcard driver to be a host app, and would appear to the host app (DAW software, etc) to be a driver with some input signals added. I had this idea after rereading the ASIO spec, so I was thinking in terms of ASIO. But I could imagine that this concept might work with WDM too. My question is - does this seem like a workable idea, or is there some obvious gotcha I hadn't thought of? Would it be much much easier to do in the ASIO domain? I'm guessing it would, but then one would be limited to devices that support ASIO. Or would it be something that could also be done in the WDM domain? I would very much appreciate any opinions on this! - Andrew Voelkel