[Libav-user] mjr to opus audio conversion - corrupted results

Talgorn François-Xavier fxtalgorn at yahoo.fr
Wed Jul 8 21:46:02 CEST 2015


I agree with you Mateusz, FFMPEG makes use of libav. If I remember well, it is built on libav + proper extensions.
It is true that FFMPEG does everything that libav does, and more, but that would be sectarian to say that this mailing list can not help you.
Technically, we can.




Le 8 juil. 2015 à 21:17, Mateusz Koslacz <mateusz.koslacz at 4vod.tv> a écrit :

> Well, so why list I'm mailing is called "Libav-user -- This list is about using libavcodec, libavformat, libavutil, libavdevice and libavfilter.", if you can't help me if I use Libav?
> 
> 2015-07-08 19:26 GMT+02:00 Paul B Mahol <onemda at gmail.com>:
> 
> Dana 8. 7. 2015. 15:05 osoba "Mateusz Koslacz" <mateusz.koslacz at 4vod.tv> napisala je:
> 
> 
> >
> > Hi,
> >
> > In our project we use janus-gateway (http://janus.conf.meetecho.com/) as a webRTC gateway and also as a stream recorder. We are on the tests stage of our project, and after very long development time we have ecountered a bug that is a blocker for whole project. After real tests of recording streams using janus we realized that audio and video are out of sync in recordings, despite of fact, that live they are perfectly in sync.
> >
> > Janus records two *.mjr files for each recording - one for video, and one for audio. Janus also provides own converter to convert this files to *.webm for video and *.opus for audio. After that, we use ffmpeg to merge this files. By recording stopwatch view on the stream we concluded, that video has correct length and speed, so most probably this is something wrong with audio recording or converting.
> >
> > Firstly, janus was using libogg for converting *.mjr audio to *.opus. Regarding described issues author decided to switch to libav audio processing. You can check how it's done in this file https://github.com/meetecho/janus-gateway/blob/pprec-opus-libav/postprocessing/pp-opus.c starting from 90 line.
> >
> > Returned data is a lil bit smaller than data obtained using libogg:
> >
> >> -rw-r--r-- 1 mateusz mateusz  17987445 lip  8 11:58 rec-2215547545-audio-nolibogg.opus
> >> -rw-r--r-- 1 mateusz mateusz 221841174 lip  8 11:56 rec-2215547545-video.webm
> >> -rw-r--r-- 1 mateusz mateusz  18074102 lip  8 11:52 rec-2215547545-audio-libogg.opus
> >
> >
> > Unfortunately, returned data is somehow corrupted - ffprobe results for libav processing output:
> >
> >> [ogg @ 0x4336540] Codec not found
> >> rec-2215547545-audio-nolibogg.opus: End of file
> >
> >
> > and for compare, for libogg version:
> >
> >> Input #0, ogg, from 'rec-2215547545-audio-libogg.opus':
> >>   Duration: 01:20:32.98, start: 0.000000, bitrate: 29 kb/s
> >>     Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
> >
> >
> > Moreover, no player can play this file - VLC just does nothing, and Rythmbox says "Can't play file: {null}".
> >
> >
> > Is this because of improper use of libav api, or maybe it is some libav bug? How could we possibly overcome this problems?
> >
> > For more details you can read our whole conversation with janus author here:
> > https://groups.google.com/forum/#!topic/meetecho-janus/lMCtBm7VbMY
> >
> >
> 
> From that conversation it is evident you are using Libav and not FFmpeg. And so we can not help you.
> 
> > Regards,
> >
> > Mateusz Koślacz
> >
> > _______________________________________________
> > Libav-user mailing list
> > Libav-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/libav-user
> >
> 
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> 
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