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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • La sauvegarde automatique de canaux SPIP

    1er avril 2010, par

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

Sur d’autres sites (2955)

  • Ffmpeg - padding/margins/offset in amix filter. Overlay conversation to music

    10 juin 2022, par user19313832

    Use filter_complex and amix to overlay conversation with music.

    


    There is a video file with music, and an audio file with a conversation.
When they start talking, the volume of the music fades. When the conversation ends, the volume of the music increases again. It works.

    


    But there is a problem that the volume of the music decreases only after they started talking. It is required that the volume of the music decrease even before they start talking, with an indent that can be specified. That is, when they have not said anything yet, the volume of the music decreases, for example, in a second when they start talking. Is there a solution ?

    


  • Ffmpeg [NULL @ 0x56390335ae80] Unable to find a suitable output format for 'listen' listen : Invalid argument

    18 février 2021, par Sowmya

    I am trying to record a live stream using ffmpeg. I am using an sdp file as input and trying to put it in a .mp3 file and i get the following error :

    


    ffmpeg listen  -loglevel debug -protocol_whitelist file,crypto,udp,rtp,tcp  -i audio.sdp -f mp3 output.mp3
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Splitting the commandline.
Reading option 'listen' ... matched as output url.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp,tcp'.
Reading option '-i' ... matched as input url with argument 'audio.sdp'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'output.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url audio.sdp.
Successfully parsed a group of options.
Opening an input file: audio.sdp.
[NULL @ 0x5639033039c0] Opening 'audio.sdp' for reading
[sdp @ 0x5639033039c0] Format sdp probed with size=2048 and score=50
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 48000
[sdp @ 0x5639033039c0] audio channels set to: 2
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 16000
[sdp @ 0x5639033039c0] audio channels set to: 1
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 32000
[sdp @ 0x5639033039c0] audio channels set to: 1
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 8000
[sdp @ 0x5639033039c0] audio channels set to: 1
[udp @ 0x56390330b3c0] end receive buffer size reported is 131072
[udp @ 0x56390330b5e0] end receive buffer size reported is 131072
[sdp @ 0x5639033039c0] setting jitter buffer size to 500
[sdp @ 0x5639033039c0] Before avformat_find_stream_info() pos: 1281 bytes read:1281 seeks:0 nb_streams:1
[sdp @ 0x5639033039c0] After avformat_find_stream_info() pos: 1281 bytes read:1281 seeks:0 frames:0
Input #0, sdp, from 'audio.sdp':
  Metadata:
    title           : -
  Duration: N/A, bitrate: N/A
    Stream #0:0, 0, 1/8000: Audio: opus, 48000 Hz, mono, fltp
Successfully opened the file.
Parsing a group of options: output url listen.
Successfully parsed a group of options.
Opening an output file: listen.
[NULL @ 0x56390335ae80] Unable to find a suitable output format for 'listen'
listen: Invalid argument
[AVIOContext @ 0x56390330c6c0] Statistics: 1281 bytes read, 0 seeks


    


    I have tried with output as .wav file too, but i get the same error, Please help me out. Thanks in advance !

    


  • Blacklist Safari 5.0.1 (533.17.8) on Snow Leopard (OS X 10.6.3 and 10.6.4), HTML5 audio issue still not fixed (but on Apple’s radar) - https://bugs.webkit.org/show_bug.cgi?id=32159#c9

    14 août 2010, par Scott Schiller

    m index.html m script/soundmanager2-jsmin.js m script/soundmanager2-nodebug-jsmin.js m script/soundmanager2.js Blacklist Safari 5.0.1 (533.17.8) on Snow Leopard (OS X 10.6.3 and 10.6.4), HTML5 audio issue still not fixed (but on Apple’s radar) - (...)

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