I'm prototyping some software which uses a PulseAudio sink/source pair which I wrote[1]. Even doing something as simple as having the audio source go into a GUI sink and the audio sink, I'm having an issue where as the program runs longer, the latency gets higher. Running for a day the latency will grow to many seconds. This causes issues for the external program generating and receiving the baseband signal, which is unaware of this latency.
I best guess of what's going on is this: the source block is connected to a PulseAudio monitor device. I do this so the audio output of other applications I'm running can be routed into GnuRadio's input. The same audio device is associated with the sink and source, so the clocks are normally synchronized. But as those other applications open and close the audio device, gaps are introduced into the monitor device, requiring GnuRadio's buffers to grow as the clock (temporarily) drift apart. So two questions: 1) Is there a way I can inspect the buffer sizes within GnuRadio to confirm my hypothesis? 2) Assuming that's what's going on, is there an adjustment I can make (perhaps to the flow graph, or to the PulseAudio blocks) to drop samples instead of buffering? [1]: https://github.com/bitglue/gr-pulseaudio
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