Client Setup - Audio Buffers
The Buffer length - as explained in the Server Audio Codecs - applies to the microphone device of the client.
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 8 to 10 but you can play with it. The lower the value, the lower the soundcard latency. You will notice little gaps with very small values. This value can always be changed at runtime.
The number of buffers to be processed in the soundcard queue can be watched by hitting the keyboard shortcut Alt+B. The number turns red if the number of buffers gets too high.
If you mute (M) the audio for a while the buffer queue goes down to zero.