Remote Audio Client - Audio Buffers
Most of the latency on the VoIP path from your radio to your ears is generated on your own computer! One major source of latency is the soundcard. The audio stream arrives in many small chunks of data. The soundcard needs time to process the data and to convert it from digital to analogue. If the audio stream packets arrive in a higher frequency than the soundcard can process them, the audio stream is buffered and the sound output delayed. This can lead to delays of 500ms or even more than one second and that is unacceptable for remote radio audio.
The soundcard buffer overrun can be limited by setting a number higher than 0. A recommended value is 6 to 10 but you can play with it. The lower the value, the lower the soundcard latency. You will notice gaps with very small values. This value can always be changed at runtime.