The sound and microphone redirection channels (and in part TSMF)
did not properly decouple encoding/decoding from the backends used
to play/record sound.
Encapsulating encoding/decoding in rewritten freerdp_dsp_* functions
with variable backends, simplifying alsa/oss/pulse/... audio backends.
The rdpContext gets an event which will
get set if an error occoured in a channel.
If a thread or a void callback has to report an
error it will get signaled by this system.
This patch adds checks for the sound channel and plugins:
* checks for protocol bytes;
* checks for malloc/calloc;
* be more strict when interacting with sound plugin
