Recherche avancée

Sur d’autres sites (6176)

• ffmpeg shutting down when stream is silent

  7 février 2017, par chr_lt_ney

  I am running ffmpeg in a Linux environment, where I merge the audio stream from an Icecast-stream and a static image in order to be able to stream that to Youtube.

  At times, the Icecast stream get’s very silent, sometimes for almost a minute. That is intended to happen, due to the nature of the content. But, when the stream get’s too silent, ffmpeg just shuts down with no notice.

  This is how I start ffmpeg :

  ffmpeg -loop 1 -r 30  -s 1280x720 -i /var/www/html/files/youtube/image.png -i http://my-stream-to-icecast:8000/my-mount -c:v libx264 -preset fast -tune stillimage -crf 18 -c:a copy -shortest -pix_fmt yuv420p -f flv rtmp://a.rtmp.youtube.com/live2/my-youtube-key

  Any ideas on how to solve this ? And/or how to optimize the above ?

  Thanks a lot !

• 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.

• Anomalie #3271 (Nouveau) : tailles_en_octets : tenir compte des norme SI

  22 septembre 2014, par Maïeul Rouquette

  On va pas revenir sur l’historique des puissance de 2 comme unité de mesure, mais actuellement taille_en_octets fait comme si 1 ko = 1024 octets.

  Depuis la normalisation par l’iso, on distingue :
  - 1 ko = 1000 octets
  - 1kio = 1024 octets

  et de même pour les multiples au dessus. Cela n’a guère d’importance pour les petites unités, mais à partir du giga cela se fait sentir.

  Voir https://fr.wikipedia.org/wiki/Octet#Multiples_normalis.C3.A9s

  Proposition pour que SPIP soit conforme aux normes ISO :
  - changer les chaînes de langues pour utiliser le kibi au lieu du kilo
  - Utiliser les multiples de 10 dans la fonction taille_en_octets, en proposant une constante pour basculer vers l’ancien mode