Hello (my apologies if this post went to wrong sub-forum, not totally sure if my problems are networking or sound-api related),
I'm currently building a kind of real-time "echo-laboratory". I'm having a tough time making the part of my code using JavaSound + plain old sockets meet the requirements - e.g. buffer size of approximately 80 bytes when using a 8 kHz sample-rate (mono, i.e. 16-bit PCM).
What I try to achieve is basically:
"rec mic" + "network send" + "network receive" + "playback on loudspeaker" <= 20 ms latency (<10 ms if possible would be great!)*
So my questions to you guru's stalking
this forum are:
* Is the above scenario realistic to solve using java with decent hardware and networking support? My guess is yes, otherwise I wouldn't bother you guys - but still, if not...
* I'm currently using Source/Target-DataLine's from the JavaSound API and have trouble with small buffer size's (i.e. low latency requirement), is it my lack of understanding (hey, I just recently begun using the api
) the api or should I consider alternative API's; if so, any suggestions?
* I'm currently using plain old sockets to do network communication (currently only "point-to-point"), and without any in-depth performance profiling that seems to work ok, i.e., mostly dependent on inherent network lags.
Thanks in advance
* Final architecture will use an "in-lab network" - network latency should not be a (big) problem (<< 10 ms round-trip).