[FFmpeg-devel] Formats with 0 channels

[Michael Niedermayer]

On Mon, Jun 02, 2008 at 09:13:09AM +0200, Andreas ?man wrote:
> Hi
>
> Some streams have the AAC channel configuration stored as a PCE
> element inside the stream, but sample rate can be recovered
> from format.
>
> This causes ffmpeg to believe that it should use 0 channels
> for output:
>
> $ ./ffmpeg_g  -i samples/aac_lc/al03_48.mp4 -f s16be /dev/null
> FFmpeg version SVN-r13618, Copyright (c) 2000-2008 Fabrice Bellard, et al.
>   configuration:
>   libavutil version: 49.7.0
>   libavcodec version: 51.57.0
>   libavformat version: 52.14.0
>   libavdevice version: 52.0.0
>   built on Jun  2 2008 08:44:21, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 
> 4.1.2-16ubuntu2)
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'samples/aac_lc/al03_48.mp4':
>   Duration: 00:00:08.23, start: 0.000000, bitrate: 67 kb/s
>     Stream #0.0(und): Audio: aac, 48000 Hz, 0 channels
> File '/dev/null' already exists. Overwrite ? [y/N] y
> Output #0, s16be, to '/dev/null':
>     Stream #0.0(und): Audio: pcm_s16be, 48000 Hz, 0 channels
> Stream mapping:
>   Stream #0.0 -> #0.0
> Press [q] to stop encoding
> Floating point exception (core dumped)
>
> The attached patch fixes this by checking for both a valid
> sample rate and number of channels, i.e:

ok

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Everything should be made as simple as possible, but not simpler.
-- Albert Einstein
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