LibAudio: Add a Serenity implementation of PlaybackStream

This implementation is very naive compared to the PulseAudio one.

Instead of using a callback implemented by the audio server connection
to push audio to the buffer, we have to poll on a timer to check when
we need to push the audio buffers. Implementing cross-process condition
variables into the audio queue class could allow us to avoid polling,
which may prove beneficial to CPU usage.

Audio timestamps will be accurate to the number of samples available,
but will count in increments of about 100ms and run ahead of the actual
audio being pushed to the device by the server.

Buffer underruns are completely ignored for now as well, since the
`AudioServer` has no way to know how many samples are actually written
in a single audio buffer.

Userland/Libraries/LibAudio/ConnectionToServer.h

@@ -54,6 +54,8 @@ public:
     // How many buffers (i.e. short sample arrays) the server hasn't played yet.
     // Non-realtime code needn't worry about this.
     size_t remaining_buffers() const;
+    // Whether there is room in the realtime audio queue for another sample buffer.
+    bool can_enqueue() const;
 
     void set_self_sample_rate(u32 sample_rate);