using ffmpeg for cutting media files – and the gotchas involved

So here I was, trying to complete some ordinary transcoding / media file cutting tasks with ffmpeg. Turns out there are some weird gotchas when using some versions of ffmpeg (in my case, SVN-r10571) on the commandline.

Problem 1: transcoding to mp3 insists on using 64kbits/s or 128kbits/s bitrate

So a simple task is to transcode an audio file of some arbitrary format to MP3. In my case, I only wanted a 30-second piece of the original file, converted to MP3. The original was in 256kbits/s, and I decided to use the same bitrate for the output, just for kicks. Reading the man page, -b is for video bitrate, and -ab is for audio bitrate. So I executed:

wrong:  ffmpeg -ss 00:00:30.00 -t 25 -i foo.mp3 -ab 256 foo-new.mp3

The output, of course, was this:

Input #0, mp3, from 'foo.mp3':
mdb:109, lastbuf:0 skipping granule 0
size= 393kB time=25.2 bitrate= 128.0kbits/s
video:0kB audio:393kB global headers:0kB muxing overhead 0.007950%

128 kbits/s. That was…not what I wanted.

Turns out, as the ffmpeg man page hints but does not specify clearly, a “k” is required to label the units of the new bitrate. As in:

right:  ffmpeg -ss 00:00:30.00 -t 25 -i foo.mp3 -ab 256k foo-new.mp3

This time, the output was:
Input #0, mp3, from 'foo.mp3':
mdb:109, lastbuf:0 skipping granule 0
size= 782kB time=25.0 bitrate= 256.0kbits/s
video:0kB audio:782kB global headers:0kB muxing overhead 0.003996%

Much better. It’s very strange how that if I don’t give it the ending “k” but do give it a higher value than 64, it always bumps up the bitrate to 128kbits/s, from the default of 64k (but not to the number I actually wanted).


Problem 2: cutting media file without re-encoding

So ffmpeg can be used to cut a media file, without reencoding the media stream. You simply pass the raw copy codec to the -acodec (or the -vcodec for video) at encode time. For my MP3, I thought it was a pretty trivial problem, so I issued:

wrong: ffmpeg -i bar.mp3 -ss 00:00:30.00 -t 25 -acodec copy bar-new.mp3

This seemed to process correctly, except it created a 555-byte empty file with no content in it. What’s even weirder, if you issued:

wrong: ffmpeg -i bar.mp3 -ss 00:00:10.00 -t 25 -acodec copy bar-new.mp3

That is, -ss 00:00:10.00 to seek to the 10th second, and -t 25 to record 25 seconds worth of audio. Strangely enough, the output file had 15 seconds of audio, the subtraction of 25 by 10. Curiouser and curiouser. Now,

wrong: ffmpeg -i bar.mp3 -ss 00:00:10.00 -t 25 -acodec mp3 bar-new.mp3

does in fact create a 25-second MP3 file. Only, of course, a re-encoded one at 64kbits/s. Something that we are not looking for. But strange.

A post on ffmpeg-devel finally provided enough hints to clue me into the problem. Unlike many command line apps, the order of arguments passed to ffmpeg seem to be silently significant. The correct incantation is:

right: ffmpeg -ss 00:00:30.00 -t 25 -i bar.mp3 -acodec copy bar-new.mp3

Note the switched order for the arguments -i and the args -ss and -t, where the -i must follow the other two. Now, the desired 25-second file, cut from the original, is correctly produced.

When arguments are missing some arbitrary text or in the wrong order, ffmpeg doesn’t sanity-check or warn you of these…it just silently proceeds and does some very strange things. Things that make you scratch your head and wonder, “wtf? did I mistype an argument somewhere?”

Ah, and of course, you make me have to track down all these little idiosyncrasies, wading through blog posts and mailing lists and forums. Agh.