2018-08-22 15:39 GMT+02:00, Amir Raza <[email protected]>: > This is my first mail to ffmpeg , apologies if make some mistakes.
User questions on the development mailing list are not welcome, as is cross-posting > I tried many online example codes for decoding aac audio to wav file. > including example codes which is for MP2 codec. > below is one such example code , it doubles the decoded file size (.wav) > but not playable. You are not writing a wav file, this is visible because you are not including libavformat header files which would be necessary to write a wav file. I suspect you are writing an f32le file which should be playable with something like: $ ffplay -f f32le -ac 2 -ar 48k filename. (Depending on the properties of the aac input file, the example is for 48kHz stereo) I thought that adts files cannot be read without libavformat but if you see an output filesize>0, this is apparently not true. Carl Eugen _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
