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  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

  • Configurer la prise en compte des langues

    15 novembre 2010, par

    Accéder à la configuration et ajouter des langues prises en compte
    Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
    De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
    Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...)

Sur d’autres sites (3162)

  • can't decode RTMP stream from adobe FMS

    25 juillet 2013, par Mike Versteeg

    I have written code to decode RTMP streams but ran into a problem decoding a stream from FMS. Same stream from Wowza server works fine, but when using Adobe FMS I
    keep getting the same error (note it works fine in a flash player).

    I can confirm the problem using ffmpeg.exe, here's the output of the latest git, anyone have an idea ?

    ffmpeg version N-54901-g55db06a Copyright (c) 2000-2013 the FFmpeg developers
     built on Jul 23 2013 18:01:29 with gcc 4.7.3 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
    ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
    ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
    eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
    amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
    enable-libxvid --enable-zlib
     libavutil      52. 40.100 / 52. 40.100
     libavcodec     55. 19.100 / 55. 19.100
     libavformat    55. 12.102 / 55. 12.102
     libavdevice    55.  3.100 / 55.  3.100
     libavfilter     3. 81.102 /  3. 81.102
     libswscale      2.  4.100 /  2.  4.100
     libswresample   0. 17.103 /  0. 17.103
     libpostproc    52.  3.100 / 52.  3.100
    Parsing...
    Parsed protocol: 0
    Parsed host    : [removed for privacy reasons]
    Parsed app     : vidlivestream/_definst_/stream
    RTMP_Connect1, ... connected, handshaking
    HandShake: Type Answer   : 03
    HandShake: Server Uptime : 506058230
    HandShake: FMS Version   : 4.5.5.1
    HandShake: Handshaking finished....
    RTMP_Connect1, handshaked
    Invoking connect
    HandleServerBW: server BW = 1250000
    HandleClientBW: client BW = 1250000 2
    HandleChangeChunkSize, received: chunk size change to 1024
    HandleCtrl, received ctrl. type: 6, len: 6
    HandleCtrl, Ping 506058630
    sending ctrl. type: 0x0007
    RTMP_ClientPacket, received: invoke 242 bytes
    (object begin)
    Property:
    Property:
    Property:
    (object begin)
    Property: 4,5,5,4013>
    Property:
    Property:
    (object end)
    Property:
    (object begin)
    Property:
    Property:
    Property:
    Property:
    Property:
    (object begin)
    Property:
    (object end)
    (object end)
    (object end)
    HandleInvoke, server invoking <_result>
    HandleInvoke, received result for method call <connect>
    sending ctrl. type: 0x0003
    Invoking createStream
    RTMP_ClientPacket, received: invoke 21 bytes
    (object begin)
    Property:
    Property:
    Property: NULL
    (object end)
    HandleInvoke, server invoking <onbwdone>
    Invoking _checkbw
    RTMP_ClientPacket, received: invoke 29 bytes
    (object begin)
    Property:
    Property:
    Property: NULL
    Property:
    (object end)
    HandleInvoke, server invoking &lt;_result>
    HandleInvoke, received result for method call <createstream>
    SendPlay, seekTime=0, stopTime=0, sending play: test
    Invoking play
    sending ctrl. type: 0x0003
    RTMP_ClientPacket, received: invoke 16419 bytes
    (object begin)
    Property:
    Property:
    Property: NULL
    Property:  K H 7 ~ + $ K Z #   ! v 1 &lt; m N % h 9 n G t % J M p 1 f # t %
    ^ u ( I ^ ) &lt; 5 : ? @ a V O &lt; S n [ * y N y T e * 3 P 1 F ! 6 #   + ( w > W \ -
    : = ` _ 6 q $ - 0 e x G . &#39; 4 [ * / 0 / &amp; _ l ] @ k 8 )v>
    Property:
    (object end)
    HandleInvoke, server invoking &lt;_onbwcheck>
    Invoking _result
    HandleChangeChunkSize, received: chunk size change to 1024
    RTMP_ClientPacket, received: invoke 142 bytes
    (object begin)
    Property:
    Property:
    Property: NULL
    Property:
    (object begin)
    Property:
    Property:
    Property:
    Property:
    (object end)
    (object end)
    HandleInvoke, server invoking <onstatus>
    HandleInvoke, onStatus: NetStream.Play.Failed
    Closing connection: NetStream.Play.Failed
    </onstatus></createstream></onbwdone></connect>

    PS : although there is some resemblance to this topic, it is very old (certainly in ffmpeg terms) and the suggestions make no difference.

  • Why audio element currentTime on ffmpeg encoded mp3 file in Chrome browser does not work

    26 juillet 2013, par Peter

    I have an HTML5 audio element :

    <audio preload="auto">
       <source src="./Sound/recording.mp3" type="audio/mpeg">
    </source></audio>

    and I need to be able to play last 4 seconds from mp3 recording. My javaScript is :

    audio.currentTime = audio.duration-4;
    audio.play();

    Works ok in IE10 and Firefox, but Chrome starts playing from a wrong place. The difference between reported audio.currentTime and actual playback position is about 20s. The recording.mp3 is created with ffmpeg :

    ffmpeg -i recording.wav -ab 32k recording.mp3

    It works, when I strip the ID3v2 header from the recording.mp3 (deleting the first couple bytes in the file before the audio data).

    It also works when I compress to ogg. Can somebody point me to the right direction (ffmpeg switches, audio element attributes or whatever) to get it work also in chrome ?

    Thanks in advance

    EDIT :
    the ffmpeg output :

    ffmpeg version N-53528-g160ea26 Copyright (c) 2000-2013 the FFmpeg developers
     built on May 27 2013 15:20:09 with gcc 4.7.3 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
    ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
    ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
    eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
    amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
    enable-libxvid --enable-zlib
     libavutil      52. 34.100 / 52. 34.100
     libavcodec     55. 12.100 / 55. 12.100
     libavformat    55.  7.100 / 55.  7.100
     libavdevice    55.  1.101 / 55.  1.101
     libavfilter     3. 72.100 /  3. 72.100
     libswscale      2.  3.100 /  2.  3.100
     libswresample   0. 17.102 /  0. 17.102
     libpostproc    52.  3.100 / 52.  3.100
    [wav @ 0433e840] max_analyze_duration 5000000 reached at 5015510 microseconds
    Guessed Channel Layout for  Input Stream #0.0 : mono
    Input #0, wav, from &#39;recording.wav&#39;:
     Duration: 02:30:07.86, bitrate: 176 kb/s
       Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, mono, s16, 176 kb/s
    Output #0, mp3, to &#39;recording.mp3&#39;:
     Metadata:
       TSSE            : Lavf55.7.100
       Stream #0:0: Audio: mp3 (libmp3lame), 11025 Hz, mono, s16p, 32 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (pcm_s16le -> libmp3lame)
    Press [q] to stop, [?] for help
    size=   35188kB time=02:30:07.86 bitrate=  32.0kbits/s
    video:0kB audio:35187kB subtitle:0 global headers:0kB muxing overhead 0.000672%
  • doc/encoders : partially rewrite and reformat libx264 docs

    23 juillet 2013, par Timothy Gu
    doc/encoders : partially rewrite and reformat libx264 docs
    

    Format is based on the thread :
    "[PATCH] doc/encoders : Add libopus encoder doc" (06-28-2013)
    http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/165368/

    Also merge the two option sections (Mapping and Private options).

    Patch partially edited by Stefano Sabatini.

    Signed-off-by : Stefano Sabatini <stefasab@gmail.com>

    • [DH] doc/encoders.texi