Hello all,

I'm encoding my audio files to opus, with the following command:
> ffmpeg -i test.mp3 -acodec libopus -b:a 64000 -vbr on test.opus
(Debian squeeze, ffmpeg version 3.3.5)

According to wikipedia, the "opus" container should just be an ogg container 
(https://en.wikipedia.org/wiki/Opus_%28audio_format%29).
However, encoding my file with an ogg extension :
> ffmpeg -i test.mp3 -acodec libopus -b:a 64000 -vbr on test.ogg
produces a different file from the previous one (md5 are mismatched).

Indeed, ffmpeg tells me so : "(...)Output #0, ogg, to 'test.ogg':(...)" instead 
of "(...)Output #0, opus, to 'za.opus':(...)"
However, the "file" util view both files as ogg :
> file test.*
test.ogg:  Ogg data, Opus audio,
test.opus: Ogg data, Opus audio,


What's going on here ?
What is the container used when encoding a file to ".opus" ?
Which one should I use (I'm aiming for maximum compatibility, mainly with 
android phones) ?

Thanks for reading !

N
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