I am trying to encode raw audio (pcm_f32le) to AAC encoded audio. One thing
I've noticed is that I can accomplish this via the CLI tool:
ffmpeg -f f32le -ar 48000 -ac 2 -c:a pcm_f32le -i out.raw out.m4a -y
This plays just fine and decodes fine.
The steps I've taken:
*
When I am using the C example code:
https://ffmpeg.org/doxygen/3.4/encode_audio_8c-example.html and switch the
encoder to codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
*
Output the various sample formats associated with AAC, it only provides FLTP.
That assumes a planar/interleaved format.
*
This page (
https://forum.videohelp.com/threads/373264-FFMpeg-List-of-working-sample-formats-per-format-and-encoder
) seems to provide the various supported input formats per codec.
This is confusing because I don't think my raw captured audio is interleaved.
I've certainly tried passing it through and it doesn't work as intended.
It will stay stuck here with this ret code indefinitely after calling
avcodec_receive_packet :
AVERROR(EAGAIN): output is not available in the current state - user must try
to send input
Questions:
*
How can I modify the example code from FFmpeg to convert pcm_f32le raw audio to
AAC encoded audio?
*
Why is the CLI tool able to?
*
I am using libsoundio ( http://libsound.io/ ) to capture raw audio from Linux's
Dummy Output. I wonder how I could get a planar format to pass through to get
AAC encoded audio.
*
If AAC is not a possibility, is doing so with MP3?
Suhail
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