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I am converting a bunch of videos from FLV to MP4 (for iphone compatibility) and using AAC codec it kept throwing out "invalid audio bitrate" or similar errors. I tried -c:a copy option and it works, but I am wondering if that tells avconv to use the original MP3 codec or just the original audio bitrates/frequencies?

avconv -i input.flv -s 640x480 -b 1248k -vcodec libx264 -acodec aac \
-strict experimental -c:a copy output.mp4
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migrated from Jun 3 '13 at 5:09

This question came from our site for professional and enthusiast programmers.

Please update your question by including your ffmpeg command and the complete ffmpeg console output. Also ffmpeg is not avconv. True ffmpeg is by the FFmpeg project and avconv is from a fork called libav. – LordNeckbeard May 31 '13 at 23:09
but they both have the -c:a copy option which is my focus on this question, so stating that ffmpeg and avconv are not the same doesnt really apply... my question regards -c:a copy for aac codec – b747fp May 31 '13 at 23:37

-c:a copy means that the input audio will be copied as is, without any transcoding. So if your input has mp3 audio, the output will also be mp3, an exact copy of the input.

The reason it complains about invalid bitrate is most probably -b 1248k. That option written like this means 'set the bitrate of all output streams to this value'. What you probably want is -b:v, which only sets the bitrate for video. Similarly, -b:a would set the bitrate for all audio streams.

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