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• run ffmpeg commands from my own project

  28 octobre 2013, par bruno

  I'm starting a project where I want ppl to upload videos of a talk and a the video of slides for that talk and want to merge them (to play at the same time) and then show results.

  My question is :
  Is it possible to do that from code ? if it is, can you point me to the right doc ?
  I was able to do it running command line, but as I want this to run on a server with different ppl uploading their videos I think this would not be the best approach.
  I have a preference for Java if it's possible to do it, but I can manage to use other languages what do you guys suggest ?

  The idea would be to have a service where I can point the urls of the videos stored in my server and it would merge them and save file where I can later stream. With different ppl uploading videos at the same time and being able to watch the result in a reasonable amount of time.

  I used this tutorial to test :
  https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20input%20videos

  Thanks for your time

• Anomalie #3523 : Gestion des plugins avec spip 3.0.20

  5 août 2015, par Gilles VINCENT

  en ajoutant la directive
  

  opcache.validate_timestamps=0

  à ma config php, comme conseillé dans http://wiki.mikejung.biz/PHP_OPcache pour un serveur en production, la page de gestion des plugins refonctionne à nouveau correctement. De plus, la valeur
  

  opcache.revalidate_freq=60

  (valeur par défaut fournie lors de l’install) est vraiment longue.
  Si je commente cette ligne dans le fichier de config, le phpinfo() indique une valeur plus raisonnable de 2 seulement.
  Je pense que le chargement méga long venait de cette constante.

  La différence de comportement avec la config "pour développement" entre une 3.0.19 et une 3.0.20 reste encore surprenante.
  Si j’ai le temps, je creuserai dans une semaine à mon retour de vacance, car tout le monde ne peut as jouer avec ces paramètres du serveur... :/

• Transcode HLS Segments individually using FFMPEG

  27 mai 2013, par rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line :

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio :

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue) :

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.