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 up to 500 ms or even more 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. This value can always be changed at runtime. You might notice little audio gaps with very small values which is acceptable for CW and SSB. If you decode WSJT modes such as JT65, FT8, FT4 on your client computer you should set this value to 0.
The number of buffers which are currently processed in the soundcard queue are shown in the right bottom corner of the application. 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.