RTP Buffering - Frame Based Buffering

Revision as of 14:18, 22 June 2010 by Brain (talk | contribs)

Introduction

In Song module version 8 a new RTP buffering method called frame based buffering was introduced. The algorithm calculates the audio buffer level in milliseconds rather than in bytes.

Features

Frame based buffering allows:

  • configurable decoding delay with one frame accuracy
  • synchronisation of several decoders to the same stream (just by configuring them to the same initial delay)
  • stable delay over long period of time
  • automatic correction of clock difference between encoder and decoder

Applications

The following applications use frame based buffering:

Application Name Version
Streaming Client 2.17
Annuncicom Full Duplex 0.21
RTP STL 2.01


Configuration