Streams are synchronized based on "chunk receival time", and not "stream time"

[blaa]

With rate, sample and channels defined, each incoming sample of data has a precise moment when it should be played.

• data_received reads incoming data and puts in a buffer, then into a queue.
• packetize() reads this data eagerly and marks with a time - based on the moment of READING, not the stream queue.

Instead, it would be better to track the STREAM TIME. This creates a problem with input underflow which will have to be handled.

This is mostly fine when the source is streaming audio in it's natural time. If the pulseaudio sends as a bigger chunk od data "to be played later" - weird things can happen. This wasn't problem when I used it with spotify/mopidy though.

[blaa]

I've implemented this approach in the "2.0" release. It seems to work in my environment, but I tested with only one client (which had huge problems before implementation and now it's certainly better in some ways)