[Libav-user] Hard time encoding AAC (works for MP3) - what settings are needed?

[Jesper Taxbøl]

I am struggling to make AAC encoding work with libav.

I have an example working for MP3. But I can't seem to figure out what is
needed to make AAC encoding work. I get  a whole bunch of errors when
playing my encoder output.

I suspect I need some extra configuration for encoder context, but I have
not been able to find hints that can help me.

I have made a repo here with my code:
https://github.com/taxfromdk/audio_encode_test.git

The juicy code is here:
https://github.com/taxfromdk/audio_encode_test/blob/master/encode_audio.cpp

It can be run by checking the project out out and run "make test"

I suspect it could be related to my workflow, so I will just note that I
test the output data by creating a file with either the extension .aac or
.mp3 where I dump the raw encoded packages and play through ffplay.

I am on ffmpeg version n4.3.1 under Arch linux.

The output from ffplay in case of the AAC file is listed below.

I would appreciate pointers to what is needed for the codec context to spit
out working AAC, as I need it to accompany some video. :)

Kind regards

Jesper

------

ffplay raw.aac -autoexit
ffplay version n4.3.1 Copyright (c) 2003-2020 the FFmpeg developers
[aac @ 0x7fe6f8000bc0] Format aac detected only with low score of 1,
misdetection possible!
[aac @ 0x7fe6f8002600] Error decoding AAC frame header.


[aac @ 0x7fe6f8002600] 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 o
ccurs, it means that your file has a feature which has not been
implemented.


[aac @ 0x7fe6f8002600] Sample rate index in program config element does not
match the sample rate index configured by the container.

[aac @ 0x7fe6f8002600] Inconsistent channel configuration.


[aac @ 0x7fe6f8002600] get_buffer() failed


[aac @ 0x7fe6f8002600] invalid band type


[aac @ 0x7fe6f8002600] channel element 2.14 is not allocated


[aac @ 0x7fe6f8002600] Sample rate index in program config element does not
match the sample rate index configured by the container.

[aac @ 0x7fe6f8002600] Too large remapped id 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 @ 0x7fe6f8002600] If you want to help, upload a sample of this file to
https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing
list. (
ffmpeg-devel at ffmpeg.org)


[aac @ 0x7fe6f8002600] Multiple frames in a packet.


[aac @ 0x7fe6f8002600] Reserved bit set.


[aac @ 0x7fe6f8002600] Number of scalefactor bands in group (58) exceeds
limit (47).

[aac @ 0x7fe6f8002600] Reserved bit set.


[aac @ 0x7fe6f8002600] Number of bands (24) exceeds limit (22).


[aac @ 0x7fe6f8002600] channel element 1.0 is not allocated


[aac @ 0x7fe6f8002600] Reserved bit set.


[aac @ 0x7fe6f8002600] Number of scalefactor bands in group (15) exceeds
limit (12).

[aac @ 0x7fe6f8002600] channel element 1.10 is not allocated


[aac @ 0x7fe6f8002600] SBR was found before the first channel element.


[aac @ 0x7fe6f8002600] channel element 3.6 is not allocated


[aac @ 0x7fe6f8000bc0] Packet corrupt (stream = 0, dts = NOPTS).


[aac @ 0x7fe6f8002600] Sample rate index in program config element does not
match the sample rate index configured by the container.

[aac @ 0x7fe6f8002600] Inconsistent channel configuration.


[aac @ 0x7fe6f8002600] get_buffer() failed


[aac @ 0x7fe6f8002600] channel element 3.14 is not allocated


[aac @ 0x7fe6f8002600] Sample rate index in program config element does not
match the sample rate index configured by the container.

[aac @ 0x7fe6f8002600] Inconsistent channel configuration.


[aac @ 0x7fe6f8002600] get_buffer() failed


[aac @ 0x7fe6f8000bc0] decoding for stream 0 failed


[aac @ 0x7fe6f8000bc0] Estimating duration from bitrate, this may be
inaccurate
[aac @ 0x7fe6f8000bc0] Could not find codec parameters for stream 0 (Audio:
aac (Main), mono, fltp, 58 kb/s): unspecified sample rate

Consider increasing the value for the 'analyzeduration' and 'probesize'
options


Input #0, aac, from 'raw.aac':


 Duration: 00:00:04.26, bitrate: 58 kb/s
   Stream #0:0: Audio: aac (Main), mono, fltp, 58 kb/s
Failed to open file 'raw.aac' or configure filtergraph
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