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

[Suhail Doshi]

On Tue, Jan 21 2020 at 5:28 AM, < ceffmpeg at gmail.com > wrote:

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> Am Di., 21. Jan. 2020 um 04:25 Uhr schrieb Suhail Doshi
> <suhail at mightyapp.com>:
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>> On Sun, Jan 19, 2020 at 4:28 PM Carl Eugen Hoyos <ceffmpeg at gmail.com>
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>>> Am Mo., 20. Jan. 2020 um 01:22 Uhr schrieb Suhail Doshi
>>> <suhail at mightyapp.com>:
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>>> > Sure, do you know why ffmpeg cli seems to be able to encode interleaved
>>> > raw audio but the C API only allows FLTP then?
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>>> It (automatically) inserts the aresample filter into the filter chain.
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>>> Please find out what top-posting means and avoid it here, Carl Eugen
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>> Got it. So, I tried to resample my FLT audio into FLTP audio as well. I
>> got a bit stuck.
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>> 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.
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>> 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.
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> 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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Yes, I did this with the same file and it works as expected: ffmpeg -f f32le -ar 48000 -ac 2 -c:a pcm_f32le -i out.raw out.m4a -y

This is essentially what I am trying to write in C/C++ and then plan to make it work on streaming data.

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> Carl Eugen
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