[FFmpeg-trac] #1693(avcodec:new): AAC Scalable Sample Rate (SSR)

[FFmpeg]

#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):

 Parsing gain control was indeed trivial. With the attached patch
 (assembled from the summer of code repo) the SSR samples in c#5 (as*.mp4)
 decode without error.

 However, it's not sufficient to decode the ssr_chrome.aac file from the
 m3u8 in c#3. That file now causes "[aac @ 0x32a5df854f80] Input buffer
 exhausted before END element found" to be emitted from here:

 https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/aacdec_template.c#L3223

 get_bits_left(gb) == -8, in this case after decode_spectrum_and_dequant().

 https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/aacdec_template.c#L1633

 Possibly this is because the file is actually damaged and incorrect.

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Ticket URL: <https://trac.ffmpeg.org/ticket/1693#comment:24>
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