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• 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

• 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.

• Is the values in avcC box in .mp4 video files affected only by FFmpeg version ?

  5 novembre 2015, par a man

  I am studying on the source identification of video files especially about those from smartphones.

  I got to know that the values in avcC box in .mp4 video files have the encoding options(h.264) which decoder must know when processing the encoded stream.

  And I guess most of the smartphone uses the customized FFmpeg to encode the raw stream. I want to know if the values in the avcC box are affected only by the version of FFmpeg(if not customized version is used).

  I didn’t delve into this but think that the libavcodec.so in FFmpeg fill the values in avcC box when doing encoding(is this right ?).

  So what I want to ask is if two different smartphones use the same libavcodec.so(even in the case whether other .so files, .apk file used for the recording, etc are different) and two video files which have the same resolution were filmed from each smartphone, do the values in avcC box the same ?
  I think this question may equal to "are the values in avcC box affected by other FFmpeg library or other layers in overall Android framework" ?

  ++ there is one more question ! Is there any case that two videos which have same resolution from the same smartphone have different values in avcC box ? (I suggest the the difference of encoding option originating from low-battery mode, execution conditions of other apps, etc and if any core developer customize FFmpeg for that.)

  It would be a great help if anyone let me know the answer !