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. _______________________________________________ Libav-user mailing list [email protected] https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
