[FFmpeg-user] SIGABRT in libavcodec when transcoding ac3 stream
Carl Eugen Hoyos
cehoyos at ag.or.at
Thu Dec 15 22:20:13 CET 2011
Roger James <roger <at> beardandsandals.co.uk> writes:
> ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav
This is known to be broken and unsupported, please see
http://ffmpeg.org/download.html for supported versions.
> Stream #0.1[0x45](eng): Audio: ac3, 44100 Hz, 5.0, s16, 448 kb/s
> Input stream #0.1 frame changed from rate:44100 fmt:s16 ch:5 to
> rate:44100 fmt:flt ch:5
> Warning, using s16 intermediate sample format for resampling
> ERR: ffmpeg failure: broken by signal SIGABRT/SIGIOT
> Please somebody correct me if I am wrong but ac3 files are fixed format
> and do not have header information saying that the stream is encoded in
This may be true or not but the (currently) fastest decoder for AC-3 sound
outputs float values that the (fastest) AC-3 encoder takes as input.
(Quality is not affected but iirc, it was argued that quality is even better
with the float implementation.)
> Is this something that is wrong in the mpeg2 container produced by
> Why is ffmpeg trying to transcode the audio stream?
That is what ffmpeg does if you don't explicitly tell it not to.
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