Kevin Mescher wrote: > Thanks for the quick response. PulseAudio looks like a very robust API, > and > the cross-platform nature makes my job MUCH easier. > Np. You shouldn't top post though, terribly annoying. > > I'm going to do a sanity check on my thinking here: > > Where each frame is a single sample from one or more channels: > > The number of "samples" (in this case, frames) = pa_get_stream_time(...) * > sampling_rate / 1000000 > Yes, but with the slight error that you do not use the return value from pa_get_stream_time(). (you use r_usec). > > Should I use PA_STREAM_INTERPOLATE_TIMING and PA_STREAM_AUTO_TIMING_UPDATE > to save on network traffic? > They are usually a good idea if you want stable latency values. > > For later development, I'd like to change the model to this: > > In the playback procedure, run pa_mainloop_iterate, and use a > pa_stream_write callback to set the "needs more data" flag, rather than > using a sample count method. If the flag is set, send more data, and > update > the waveform marker. > > When loading data, I would send (325 + msec_latency) msec worth of > data. Or > is there a way to calculate the optimum chunk size to send? > I'd suggest using a pull model. You tell pulse to use a buffer size of 325 ms, then give it as much data as it asks for (via the callbacks). Rgds -- Pierre Ossman OpenSource-based Thin Client Technology System Developer Telephone: +46-13-21 46 00 Cendio AB Web: http://www.cendio.com -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 251 bytes Desc: OpenPGP digital signature URL: <http://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20070214/039351da/attachment.pgp>
