#1693: AAC Scalable Sample Rate (SSR)
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             Reporter:  cehoyos     |                    Owner:
                 Type:  defect      |                   Status:  new
             Priority:  normal      |                Component:  avcodec
              Version:  git-master  |               Resolution:
             Keywords:  aac         |               Blocked By:
             Blocking:              |  Reproduced by developer:  0
Analyzed by developer:  0           |
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Comment (by dalecurtis):

 There's more samples and discussion here too,
 https://bugs.chromium.org/p/chromium/issues/detail?id=808064

 This became in issue in Chrome because we stopped silently dropping these
 packets and instead emit an error now. Not decoding the SSR packet seems
 to lead to an accumulation of ~1 seconds of audio loss every 10 seconds.

 Given the volume of complaints quite a few folks have been silently losing
 their audio over the years. The workaround for now is to re-encode with
 libfdk_aac which can parse the SSR packets:

 ./ffmpeg -acodec libfdk_aac -i <file> -acodec libfdk_aac -vcodec copy <out
 file>

 Here's a recent log from a sample I have:

 $ ffmpeg -i ssr_chrome.aac out.wav
 ffmpeg version N-89956-gcaa4bd7a9f Copyright (c) 2000-2018 the FFmpeg
 developers
   built with clang version 6.0.0 (trunk 321529)
   configuration: --enable-shared --enable-nonfree --enable-gpl --cc=clang
 --ld=clang --enable-libfdk-aac
   libavutil      56.  7.100 / 56.  7.100
   libavcodec     58.  9.100 / 58.  9.100
   libavformat    58.  7.100 / 58.  7.100
   libavdevice    58.  0.101 / 58.  0.101
   libavfilter     7. 11.101 /  7. 11.101
   libswscale      5.  0.101 /  5.  0.101
   libswresample   3.  0.101 /  3.  0.101
   libpostproc    55.  0.100 / 55.  0.100
 [aac @ 0x24ad480] Estimating duration from bitrate, this may be inaccurate
 Input #0, aac, from 'ssr_chrome.aac':
   Duration: 00:00:19.50, bitrate: 134 kb/s
     Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 134 kb/s
 File 'out.wav' already exists. Overwrite ? [y/N] y
 Stream mapping:
   Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
 Press [q] to stop, [?] for help
 Output #0, wav, to 'out.wav':
   Metadata:
     ISFT            : Lavf58.7.100
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
 stereo, s16, 1536 kb/s
     Metadata:
       encoder         : Lavc58.9.100 pcm_s16le
 [aac @ 0x24d0b00] SSR 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 @ 0x24d0b00] If you want to help, upload a sample of this file to
 ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing
 list. (ffmpeg-de...@ffmpeg.org)
 Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches
 welcome
 [aac @ 0x24d0b00] SSR 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 @ 0x24d0b00] If you want to help, upload a sample of this file to
 ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing
 list. (ffmpeg-de...@ffmpeg.org)

--
Ticket URL: <https://trac.ffmpeg.org/ticket/1693#comment:9>
FFmpeg <https://ffmpeg.org>
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