[Libav-user] Audio resampling, using libswresample or libavfilter?

[Mit Shan]

Now I am decoding and playing AAC audio file, where ffplay.c is my main
reference source. Since ffplay using SDL2 to actually play audio, it uses
"AUDIO_S16SYS" type, which represents "S16" audio sample. But audio
samples decoded from AAC is format "FLTP", so resample is needed. But
I traced ffplay and found that swresample is actually not used, instead,
it uses some libavfilter functions, which I am not familiar with.

So, here is my question: What should I do to resample the "FLTP" format
to "S16" format? Using what ffplay is doing, or just using swresample
function, what ffmpeg example "resampling_audio.c" demostrates?


If I misunderstood ffplay, just point it out.