Hello,
I am quite new to ALSA and am having a bit of trouble with underruns. I am working on a voice communication system via udp sockets and am getting underruns on receiving end very sporadically (capture end is fine). Have done a great deal of reading and searching on reasons for this but am not sure what approach is best to limit the underruns. I must not be transferring the data quickly enough over the network before the ring buffer runs out of data, which makes sense, but how would you accommodate for this? I am able to buffer a large amount of network data and then playback successfully, but obviously the delay induced is so significant that it makes real-time communication impossible. If I just recover/prepare the stream after an underrun the audio is horribly distorted. Is there a normal way of dealing with this issue that I am unable to find?
I would appreciate any insight, guidance, references for any help you can give me. Thank You.