rework the sound capture & playback model

Now uses two threads for in the playing loop
The first one reads from the capture device (ie, a phys. capture, or snd_aloop) and writes
data to a circular buffer. The second one gets data from the circular buffer and outputs
it to the playback. This model solves a lot of correlated timing bugs between read & write
tasks.
The read tasks only wakes up the write task when the buffer is 80% full.
The buffer size big enough to hold 2 seconds of sound.

The mute implementation has also been simplified, since it has been found out that it was possible
to recover from an interrupted read, by calling snd_pcm_start additionnally to snd_pcm_prepare.
Thus, the mute code consists in listening to an extra file descriptor in the read loop.
Reading from that descriptor gives the mute or unmute command sent at higher level (in the
PCM control event callback).
When a 'mute' order is get, the capture sound fd is simply backup and replaced by '-1' in the
set of the poll of the read task.
When a 'unmute' order is get, the fd is simply restored.

The start threshold is only computed for capture, and hardcoded to 1 for playback.
This removes most of the remaining EPIPE on playback.
The stop threshold has been removed. It had bad side effects on the amount of writeable
data returned by snd_pcm_avail_update (was returning too small chunks)

Signed-off-by: Thierry Bultel <thierry.bultel@iot.bzh>

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