I writing a Windows application, It receives audio data from an Android app, I use UDP to transfer data over LAN, and use RtAudio to play audio-stream.
Every UDP package payload is a audio sample array, in 32k/16bit/pcm format.
When data size is 576 bytes, 288 samples in other words, every thing is OK, we can hear a clear voice.
But when data size in 192 bytes, 96 samples in other words, the sound is not clear.
Does anyone have the problem?
It is a balancing act to determine optimum size of each buffer packet ... too large and you progressively move away from real time response yet too small and the code spends proportionately too much time negotiating the boilerplate plumbing of simply transferring the data. Looks like you have hit this lower boundary when as you say 192 bytes starts acting up.
This is true independent of transport mechanism. Also keep in mind the wall clock duration consumed when listening to a few hundred bytes is tiny (typically 42,100 samples per second for CD quality mono audio) so you will not loose much in the real time aspect to give yourself more than that lower bound you have hit.
