Questions about the audio sink clock matching algorithm

[dv]

Hello,

I am using GStreamer and RTP with one sender, and N receivers. The 
receivers all play synchronously. So far I am very happy with the results.
However, I noticed that when the clock drift tolerance is reached, and 
GST_BASE_AUDIO_SINK_SLAVE_SKEW is used as the clock matching
algorithm in the audio sink, it skews the playout pointer instantly. 
While this ensures that the playback remains synchronous, it results
sometimes in an audible blip. It would probably be better if the playout 
pointer isn't skewed instantly, but gradually over time, to mask the
blips. Is such a thing possible?