2018-03-13 17:16 GMT+01:00, Michael IV <[email protected]>: > Hi. I have a PCM 16 bit (signed) stream,which I want to encode using AAC > codec. > > Now I am following the setup of audio encoding example and where I check > for > SAMPLE FMT support by the codec,the check fails: > > while (*p != AV_SAMPLE_FMT_NONE) > { > if (*p == c->sample_fmt) > { > break; > } > p++; > } > > Because AAC codec which I opened contains only AV_SAMPLE_FMT_FLTP > > Does it mean I can't use audio data as PCM16 to encode with AAC?
Yes, either use the aresample filter to convert from pcm_s16le to pcm_flt or use libswresample directly. Carl Eugen _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
