How to use ffmpeg to encode ogg audio files?

I want to convert the following simple oggenc command to use ffmpeg instead:

oggenc -b 8 input.wav -o out.ogg

From the ffmpeg libvorbis wrapper doc, it says,

The following options are supported by the libvorbis wrapper. The oggenc-equivalent of the options are listed in parentheses.

 b (-b)

Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.

But I just don't know how to apply it to ffmpeg, I've tried,

ffmpeg -i input.aac -c:a libvorbis -b 8 out.ogg
ffmpeg -i input.aac -c:a libvorbis b 8 out.ogg
ffmpeg -i input.aac -c:a libvorbis=b:8 out.ogg

but none is working as expected.

UPDATE: What I want to know is how to "translate" the options listed in ffmpeg (libvorbis wrapper) doc into ffmpeg command. I can get away with -ab switch to do the transcode, but I don't think it is the libvorbis wrapper specific options. So if you provide the answer, please provide the demo specifying all the following options as well.

b (-b)
q (-q)
minrate (-m)
maxrate (-M)
iblock

3

2 Answers

The answer is in your question:

oggenc option corresponding ffmpeg option
-b -b:a
-q -q:a
-m -minrate
-M -maxrate

Example command

ffmpeg -i input.aac -c:a libvorbis -b:a 8k output.oga

If your desired value of 8k fails then use a higher bitrate or lower your audio sampling rate with -ar, such as -ar 8000.

Encoder and muxer info

You can see additional info and the single private option that is specific to this encoder with:

ffmpeg -h encoder=libvorbis

You can also get info about the muxer/output container format:

ffmpeg -h muxer=ogg

3

This works for me:

ffmpeg -i test.aac -c:a libvorbis -b:a 64k test.ogg

I don't think 8kb/s is a valid value, but I could be wrong. :-)

1

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