I've got an older application that originally wrote directly to the Solaris audio hardware. I'm trying to port it to using a Networked audio server. I got some preliminary results from Esound, but it doesn't work that well, and I need more support. But here's my issue:
The application sent data synchronously to the audio device, and used the sample count (accessed via Solaris' info.play.samples) to determine where to put the marker on its waveform display. It was pretty simple, and the logic went: if there are less than 1000 samples remaining to be played (samples sent - the value of info.play.samples), send more data. When I modified the code to use Esound, I returned 0 to the routine asking how many samples were still in the buffer. The marker was running WAY too fast. I'm guessing the marker on the waveform is tied to how often data is sent out. I'd like to make as little code change as possible to the application. Can anybody suggest a solution?
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