[Libav-user] How do you encode raw pcm_f32le audio to AAC encoded audio with FFmpeg (C/C++)?
Carl Eugen Hoyos
ceffmpeg at gmail.com
Tue Jan 21 15:28:23 EET 2020
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?
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