[FFmpeg-user] SIGABRT in libavcodec when transcoding ac3 stream

[Carl Eugen Hoyos]

Roger James <roger <at> beardandsandals.co.uk> writes:

> ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav 
> developers

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 
> s16.

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 
> avidemux?

Definitely not.

> Why is ffmpeg trying to transcode the audio stream?

That is what ffmpeg does if you don't explicitly tell it not to.

Carl Eugen