Earlier quoted context omitted.
Can you please elaborate further? I think PulseAudio's timer based scheduling is a big improvement over interrupt driven one. Bluetooth audio just works, per application volume control, audio streaming and switching audio sink is painless, you can even configure your resampling algorithm. Even if you use pure ALSA you still have to use a sound mixing server (dmix) since most sound cards these days lack hw mixing. Oka…
It breaks on systems with asynchronous drivers. (E.g. recording timed differently from playback) Those are allowed by ALSA APIs and are present in quite a few high profile embedded Linux devices.
I'm not familiar with this, and google isn't helping. Do you mean the time between samples, during recording, isn't fixed?