I have created a MediaStreamSource to decode an live internet audio stream and pass it to the BackgroundAudioPlayer. This now works very well on the device. However I would now like to implement some form of buffering control. Currently all works well over WLAN - however i fear that in live situations over mobile operator networks that there will be a lot of cutting in an out in the stream. What I would like to find out is if anybody has any advice on how best to implement buffering.
Does the background audio player itself build up some sort of buffer before it begings to play and if so can the size of this be increased if necessary?
Is there something I can set whilst sampling to help with buffering or do i simply need to implement a kind of storeage buffer as i retrieve the stream from the network and build up a substantial reserve in this before sampling.
What approach have others taken to this problem? Thanks, Brian
One approach to this that I've seen is to have two processes managing the stream. The first gets the stream and writes it a series of sequentially numbered files in Isolated Storage. The second reads the files and plays them.
Obviously that's a very simplified description but hopefully you get the idea.