[Libav-user] Decoing problem for aac to wav using api's code
Carl Eugen Hoyos
ceffmpeg at gmail.com
Wed Aug 22 17:54:04 EEST 2018
2018-08-22 15:39 GMT+02:00, Amir Raza <mdamirraza at gmail.com>:
> 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
$ 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.
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