[Libav-user] Audio frames and resampling

[Michael IV]

Well I cracked this one. The audio  encoding example is not a good fit for
formats like aac. One has to use also output format and write container for
this thing to work as expected. Raw aac stream is useless.

On Mar 14, 2018 15:42, "Michael IV" <explomaster at gmail.com> wrote:

> 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
>
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> 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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>>
>
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