The first basic idea of the glitch-free playback model (a
better, less marketingy name is probably timer-based audio
scheduling which is the term I internally use in the PA codebase)
is to no longer depend on sound card interrupts to schedule audio but
use system timers instead. System timers are far more flexible then
the fragment-based sound card timers. They can be reconfigured at any
time, and have a granularity that is independant from any buffer
metrics of the sound card. The second basic idea is to use playback
buffers that are as large as possible, up to a limit of 2s or 5s. The
third basic idea is to allow rewriting of the hardware buffer at any
time. This allows instant reaction on user-input (i.e. pause/seek
requests in your music player, or instant event sounds) although the
huge latency imposed by the hardware playback buffer would suggest
otherwise.
