* From 0.0.1 to 0.0.2
- Project name changed from ARTS to APERTS to avoid confusion with the
  Analog Real-Time Simulator project.
- Fixed pause computation and added getTimeUsed for RTPThread. 
- Improved frame time management of RTP send service thread.
- Network byte ordering of RTP fields.
* From 0.0.2 to 0.0.3
- Changed sampling support organization so derived objects can now
  control audio sampling correctly.
- Added sampling format as an integral part of audio codec support.
- Changed RTP timestamp update to match payload format rather than
  sampling format.
- Now used APE/config.h.
* From 0.0.3 to 0.1.0
- Major rewrite of entire stack class structure.  This was done
  to better seperate RTP components and to make better re-use of
  service threads on non RTP specific streaming protocols (such as VAT).
* From 0.1.0 to 0.1.1
- Much improved and clearer handling of receive sessions including new
  RTPOutput() request and frame timing in receive service thread.
- New RTSWait() request for timed wait on RTP delivery.
- Fixed RTSRecv() routines so that the RTP header is actually received
  from the current packet before assigning codec.
- Changed receive handling thread and RTSRecv to track the current
  effective audio digitization format in use for the current frame.
- Changed RTSPost() to pass the current in effect audio digitization.
- Changed timeout into private inherited member of base RTSThread.
- Changed RTPTransceiver so that it is now inherited from RTPReceiver
  and thereby involves less code.
* From 0.1.1 to 0.1.2
- rtp socket now has it's own frame timeout information in the form of
  frame sample size so that Write() can auto-increment the timestamp.
- Skip() now used to increment the timestamp when skipping transmission
  of idle or silent frames.
- receiver service thread updated for "timeslip" at startup to allow
  the receiver to become saturated before drawing output as needed.
  This is part of the new ServiceDelay capability of RTSThread.
- RTSPost is now called with an offset timestamp so that it can
  determine where or if the currently received real-time data packet
  can be placed in a pending buffer queue for RTSOutput.

