aac - Malformed AAC bitstream - incorrectly fixed with -bsf:a aac_adtstoasc' option

[feelart]

Observing some improper fixing on a bunch of aac using FFmeg

F:\Youtube Extract\Music>ffmpeg -i alice.aac -c:a copy aliceFF.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 12 2013 16:31:48 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --
enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enab
le-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --e
nable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-lib
theora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.101 /  3. 80.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 0000000002686fe0] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'alice.aac':
  Metadata:
    title           : Alice Deejay - Better Off Alon
  Duration: 00:03:46.41, bitrate: 77 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 77 kb/s
Output #0, mp4, to 'aliceFF.mp4':
  Metadata:
    title           : Alice Deejay - Better Off Alon
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, 77 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 000000000030a620] Malformed AAC bitstream detected: use audio bitstream filter 'aac_adtstoasc' to fix it ('-bsf:a aac
_adtstoasc' option with ffmpeg)
av_interleaved_write_frame(): Operation not permitted

Now trying, to fix it with -bsf:a aac_adtstoasc

F:\Youtube Extract\Music>ffmpeg -i alice.aac -c:a copy -bsf:a aac_adtstoasc aliceFF_bsf.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 12 2013 16:31:48 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --
enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enab
le-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --e
nable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-lib
theora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.101 /  3. 80.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 00000000002db740] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'alice.aac':
  Metadata:
    title           : Alice Deejay - Better Off Alon
  Duration: 00:03:46.41, bitrate: 77 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 77 kb/s
Output #0, mp4, to 'aliceFF_bsf.mp4':
  Metadata:
    title           : Alice Deejay - Better Off Alon
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, 77 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    2123kB time=00:02:39.17 bitrate= 109.3kbits/s
video:0kB audio:2142kB subtitle:0 global headers:0kB muxing overhead -0.899321%

However, one sees that the fix is incorrect

F:\Youtube Extract\Music>aacgain -a -k aliceFF_bsf.mp4
aliceFF_bsf.mp4
Error: invalid file format aliceFF_bsf.mp4, code=15
aliceFF_bsf.mp4 is not a valid mp4/m4a file.

FYI: ​https://github.com/elfchief/mp3gain/tree/master/aacgain

On some other aac, I see aacgain error code=6

Finally, I tried to re-encode, but then raises another error, but still no proper fix.

F:\Youtube Extract\Music>ffmpeg -i alice.aac -bsf:a aac_adtstoasc aliceFF_NoCopy_bsf.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 12 2013 16:31:48 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --
enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enab
le-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --e
nable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-lib
theora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.101 /  3. 80.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 000000000031d6c0] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'alice.aac':
  Metadata:
    title           : Alice Deejay - Better Off Alon
  Duration: 00:03:46.41, bitrate: 77 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 77 kb/s
Output #0, mp4, to 'aliceFF_NoCopy_bsf.mp4':
  Metadata:
    title           : Alice Deejay - Better Off Alon
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> libvo_aacenc)
Press [q] to stop, [?] for help
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 000000000031a280] channel element 2.0 is not allocated
Error while decoding stream #0:0: Error number -1 occurred
size=    2515kB time=00:02:39.13 bitrate= 129.5kbits/s
video:0kB audio:2487kB subtitle:0 global headers:0kB muxing overhead 1.109339%

Thanks

[feelart]

Of course stream is incomplete and stops at 2:49 and not at 3:46 as its duration.
But it just shows, the effect on a cut file.

[Carl Eugen Hoyos]

Allowing to remux the file is easy, see attached patch (poc). The problem is that neither WMP nor QT play the output file indicating that FFmpeg is correct in refusing to mux such samples.
Does MP4Box produce a file that can be decoded with one of them?

[Carl Eugen Hoyos]

Replying to feelart:

However, one sees that the fix is incorrect

F:\Youtube Extract\Music>aacgain -a -k aliceFF_bsf.mp4
aliceFF_bsf.mp4
Error: invalid file format aliceFF_bsf.mp4, code=15
aliceFF_bsf.mp4 is not a valid mp4/m4a file.

Sorry, I originally missed this part of your ticket.
I tested the output of ffmpeg -i Alice.aac -acodec copy -bsf aac_adtstoasc out.mov with the following applications:
WMP, QT, faad, totem, firefox (more than half of them certainly do not depend on FFmpeg)
All of them play the output file fine. Are you sure this is no problem with aacgain that you should report at their bugtracker?

[feelart]

Regarding MP4Box, I don't know this tool.

​https://github.com/elfchief/mp3gain/blob/master/aacgain/aacgain.cpp

Line 317&318

//ignore error 4 (scalefactor out of range) which seems to happen on some tracks
//other errors are fatal

aacgain starts to process the file almost till the end, then stops on fatal error.

Same kind of fatal error both with

ffmpeg -i alice.aac -bsf:a aac_adtstoasc -vol 312 test.aac
or
ffmpeg -y -i alice.aac -bsf:a aac_adtstoasc -af volume=1.5 test.aac

F:\Youtube Extract\Music>ffmpeg -y -i alice.aac -bsf:a aac_adtstoasc -af volume=1.5 test.aac
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 12 2013 16:31:48 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --en
able-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-l
ibgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-
libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora -
-enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-
libxavs --enable-libxvid --enable-zlib
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.101 /  3. 80.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 0000000002529fe0] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'alice.aac':
  Metadata:
    title           : Alice Deejay - Better Off Alon
  Duration: 00:03:46.41, bitrate: 77 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 77 kb/s
Output #0, adts, to 'test.aac':
  Metadata:
    title           : Alice Deejay - Better Off Alon
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: aac (libvo_aacenc), 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> libvo_aacenc)
Press [q] to stop, [?] for help
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0000000002536ea0] channel element 2.0 is not allocated
Error while decoding stream #0:0: Error number -1 occurred
size=    2440kB time=00:02:39.13 bitrate= 125.6kbits/s
video:0kB audio:2487kB subtitle:0 global headers:0kB muxing overhead -1.884090%

So far as I understand, it's a bug for ffmpeg.
I have other samples, which throw other error.

P.S. Work in progress against annoying important de-synch bugs such as ticket 2400?

[Carl Eugen Hoyos]

Replying to feelart:

I will close this ticket since I don't see much indication that there is anything that should (or can) be fixed in FFmpeg, feel free to reopen if you can add information that indicates I am wrong.

Same kind of fatal error both with

ffmpeg -i alice.aac -bsf:a aac_adtstoasc -vol 312 test.aac
or
ffmpeg -y -i alice.aac -bsf:a aac_adtstoasc -af volume=1.5 test.aac

bitstreamfilters (-bsf) only work with -codec copy so these command lines make no sense.

decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0000000002536ea0] channel element 2.0 is not allocated
Error while decoding stream #0:0: Error number -1 occurred

I assume this is what you mean with fatal errors:
The messages are shown because - as you already indicated in comment:1 - the sample was cut in the middle of the file, since it wasn't cut on a frame boundary ffmpeg inevitably will fail on the last, incomplete frame.
There is nothing fatal about these messages though, ffmpeg finishes transcoding as expected or do I miss something?

So far as I understand, it's a bug for ffmpeg.

I am not sure I understand: Did you test the output file with any professional or proprietary software or a stream analyzer? Did any of them indicate an invalid output file?

I have other samples, which throw other error.

It is (obviously) not always a bug if FFmpeg throws an error. If you think it is a bug that FFmpeg throws an error, please open a ticket on this bug tracker, if you are not sure, please post a question on the ffmpeg-user mailing list.

P.S. Work in progress against annoying important de-synch bugs such as ticket #2400?

Patch welcome but note that I suspect it is trivial to workaround that issue, just copy audio into an adts container.

[Stefano Sabatini]

Replying to cehoyos:

Replying to feelart:

I will close this ticket since I don't see much indication that there is anything that should (or can) be fixed in FFmpeg, feel free to reopen if you can add information that indicates I am wrong.

Same kind of fatal error both with

ffmpeg -i alice.aac -bsf:a aac_adtstoasc -vol 312 test.aac
or
ffmpeg -y -i alice.aac -bsf:a aac_adtstoasc -af volume=1.5 test.aac

bitstreamfilters (-bsf) only work with -codec copy so these command lines make no sense.

That's not correct, bitstream filters are applied *after* filtering and *before* sending the packet to the muxer.

[Carl Eugen Hoyos]

Replying to saste:

Replying to cehoyos:

Replying to feelart:

I will close this ticket since I don't see much indication that there is anything that should (or can) be fixed in FFmpeg, feel free to reopen if you can add information that indicates I am wrong.

Do you also disagree with the above? I absolutely may miss something but I was unable to reproduce any problems with the original sample and the original command lines.

Same kind of fatal error both with

ffmpeg -i alice.aac -bsf:a aac_adtstoasc -vol 312 test.aac
or
ffmpeg -y -i alice.aac -bsf:a aac_adtstoasc -af volume=1.5 test.aac

bitstreamfilters (-bsf) only work with -codec copy so these command lines make no sense.

That's not correct, bitstream filters are applied *after* filtering and *before* sending the packet to the muxer.

(I missed that the bitstream filters also work without -codec copy but note that for the -codec copy case there is no filtering, so your description does not apply for the typical use-case.)

Do the command lines actually make sense? I thought the aac_adtstoasc bitstream filter is only useful when stream-copying from adts to mov.

[feelart]

I will close this ticket since I don't see much indication that there is anything that should (or can) be fixed in FFmpeg

Wrong, if you try to change the volume with FFMpeg the output audio can not be read any more ! AACgain just fails to perform the volume change(normalization), while ffmpeg produces a wrong audio.

bitstreamfilters (-bsf) only work with -codec copy so these command lines make no sense.

ffmpeg -i alice.aac -c:a copy -bsf:a aac_adtstoasc -vol 312 test.aac

That does not work, because copy and vol are mutually exclusive...

Also, wrong output with

ffmpeg -i alice.aac -vol 312 test.aac

Do the command lines actually make sense?

As far, as I understand FFmpeg, to me yes, maybe not to you.

P.S. Work in progress against annoying important de-synch bugs such as ticket #2400?

Patch welcome but note that I suspect it is trivial to workaround that issue, just copy audio into an adts container.

Well, if it is that trivial, I'm surprised it's not already patched, maybe for next commit.

[Carl Eugen Hoyos]

Replying to feelart:

Also, wrong output with

ffmpeg -i alice.aac -vol 312 test.aac

Could you elaborate? The command succeeds here, what do you mean with wrong output?

P.S. Work in progress against annoying important de-synch bugs such as ticket #2400?

Patch welcome but note that I suspect it is trivial to workaround that issue, just copy audio into an adts container.

Well, if it is that trivial, I'm surprised it's not already patched, maybe for next commit.

I don't know how difficult it is to fix the issue, I wrote that it is trivial to workaround it (just remux to aac).

[feelart]

ffmpeg -i alice.aac -vol 312 test.aac
The command succeeds here.

Yes, but can't play output audio, it's rubbish.

However, converting this output to mp3, now plays well

ffmpeg -i test.aac testMP3.mp3

FYI: directly re-encode as mp3, no encoding or play problem.

ffmpeg -i alice.aac -vol 312 outMP3.mp3

CCl: mp3 is failure proofed, not aac.

My guess are that with aac encoding, there are some overflow un/miss-handling and/or un/miss properly closed output aac stream.

[Carl Eugen Hoyos]

I used the following command line:
$ ffmpeg -i Alice.aac -vol 312 -acodec aac -strict -2 test.aac
and tested the output file with faad, WMP and QT (all not FFmpeg-based), all of them decode the file without problems (as do mplayer -ac ffaac and mplayer -ac faad and vlc and ffplay).

[feelart]

Indeed, adding "-acodec aac -strict -2" to the command line makes it playable, otherwise not.

But interestingly,

ffmpeg -i Alice.aac -c:a copy -bsf:a aac_adtstoasc tMP4.mp4

Ouput plays, but aacgain can't normalize it.

While,

ffmpeg -t 159 -i Alice.aac -c:a copy -bsf:a aac_adtstoasc tMP4.mp4

159 sec being the true duration, then aacgain do normalize tMP4.mp4
If you put - t 160 or more, again aacgain can't normalize it.

So something is wrong, guessing in closing stream, either by

ffmpeg -i Alice.aac -c:a copy -bsf:a aac_adtstoasc tMP4.mp4
or
ffmpeg -t 160 -i Alice.aac -c:a copy -bsf:a aac_adtstoasc tMP4.mp4

[feelart]

To summarize, you can't copy the aac stream without forcing to specify its duration!

ffmpeg -y -t 160 -i Alice.aac -c:a copy -bsf:a aac_adtstoasc t160MP4.mp4

ffmpeg -y -i Alice.aac -c:a copy -bsf:a aac_adtstoasc tMP4.mp4
ffmpeg -t 159 -i tMP4.mp4 -c:a copy tForcedTo159MP4.mp4

ffmpeg -t 159 -i Alice.aac -c:a copy -bsf:a aac_adtstoasc t159.mp4

Aacgain, can not normalize both t160MP4 and tMP4.mp4, while both tForcedTo159MP4.mp4 and t159.mp4 do not raise any issue.

[Carl Eugen Hoyos]

Does aacgain work with the original file Alice.aac?

[feelart]

Wheter you don't want to see the bug or you consider it as non implemented feature, you've the final word and won't put the matter futher.

Try also to append/concat to another aac, you'll see it won't work

But I know, such type of issue wouldn't exist with mp3, as it is not an experimental lib.

[Carl Eugen Hoyos]

I remember having spent some time with closely related issues before you opened this ticket and re-reading the ticket, it appears I have done quite extensive testing with the sample you provided.

As explained, it appears to me that:

• FFmpeg refuses to mux files that cannot be played with certain third party application (and prints an appropriate message)
• If the right bitstream filter was selected, FFmpeg produces an output file that plays with the mentioned applications (WMP, QT etc.)
• If the input file was badly edited (does not end at a frame boundary), FFmpeg prints a warning but succeeds muxing.

There absolutely may be an issue, but I was unable to reproduce it (using the command lines you provided).

If you believe there is an issue, please explain how it can be reproduced but if the explanation is just "WMP and QT play the file but aacgain fails" than I honestly wonder why you think the bug is in FFmpeg.