[Libav-user] Audio frames and resampling

[Michael IV]

Just a dumb question regarding encode_audio.c sample in FFMPEG repo.
Should the encoded audio file be playable without container?
I am encoding with AAC,using the example code from that sample but the
resulting file
is not playable not in VLC not anywhere else.And here is what ffprobe is
telling me:



[aac @ 000000000265ac00] Format aac detected only with low score of 1,
misdetection possible!
[aac @ 00000000026dca80] Error decoding AAC frame header.
[aac @ 00000000026dca80] More than one AAC RDB per ADTS frame is not
implemented. Update your FFmpeg version to the newest one from Git. If the
problem still occurs, it means that your file has a feature which has not
been implemented.
[aac @ 00000000026dca80] Multiple frames in a packet.
[aac @ 000000000265ac00] decoding for stream 0 failed
[aac @ 000000000265ac00] Estimating duration from bitrate, this may be
inaccurate
[aac @ 000000000265ac00] Could not find codec parameters for stream 0
(Audio: aac (SSR), 0 channels, fltp, 164 kb/s): unspecified sample rate
Consider increasing the value for the 'analyzeduration' and 'probesize'
options













On Wed, Mar 14, 2018 at 2:15 PM Carl Eugen Hoyos <ceffmpeg at gmail.com> wrote:

> 2018-03-14 13:13 GMT+01:00, Michael IV <explomaster at gmail.com>:
> > So I am using that lib as you can see.
>
> Yes, sorry...
>
> Carl Eugen
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