I've been investigating RTSP and HTTP audio streaming clients and have one major issue thus far: The latency on the audio is quite severe (typically on the order of 10 to 30 seconds). Minimizing the buffer size in order to reduce the latency will produce new issues such as data loss (and crashes my test app too :-( ) Anyone know of some other way of minimizing this audio latency (perhaps even using some other protocol)? I looked into RTP (it's UDP-based) but so far there doesn't seem to be any official Android support for that - at least no working code samples. If anyone has any insight, pls share! Thnx.
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