[Libav-user] How do you encode raw pcm_f32le audio to AAC encoded audio with FFmpeg (C/C++)?

[Carl Eugen Hoyos]

Am Di., 21. Jan. 2020 um 04:25 Uhr schrieb Suhail Doshi <suhail at mightyapp.com>:
>
> On Sun, Jan 19, 2020 at 4:28 PM Carl Eugen Hoyos <ceffmpeg at gmail.com> wrote:
>>
>> Am Mo., 20. Jan. 2020 um 01:22 Uhr schrieb Suhail Doshi <suhail at mightyapp.com>:
>>
>> > Sure, do you know why ffmpeg cli seems to be able to encode interleaved
>> > raw audio but the C API only allows FLTP then?
>>
>> It (automatically) inserts the aresample filter into the filter chain.
>>
>> Please find out what top-posting means and avoid it here, Carl Eugen
>
>
> Got it.  So, I tried to resample my FLT audio into FLTP audio as well. I got a bit stuck.
>
> Here's my code: https://gist.github.com/Suhail/151e41f3eb226504c7cbd3b46c15729c
> (I didn't want to paste it here since it's long where I referenced this code heavily.
>
> What I do, as a test, is I read an entire PCM raw audio file into a buffer and then send that to the encoder.

Why don't you use a wav file and read that with libavformat / did you
test reading the pcm raw file with ffmpeg?

Carl Eugen