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  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • Soumettre améliorations et plugins supplémentaires

    10 avril 2011

    Si vous avez développé une nouvelle extension permettant d’ajouter une ou plusieurs fonctionnalités utiles à MediaSPIP, faites le nous savoir et son intégration dans la distribution officielle sera envisagée.
    Vous pouvez utiliser la liste de discussion de développement afin de le faire savoir ou demander de l’aide quant à la réalisation de ce plugin. MediaSPIP étant basé sur SPIP, il est également possible d’utiliser le liste de discussion SPIP-zone de SPIP pour (...)

Sur d’autres sites (3813)

  • Anomalie #4128 : Bug de génération de boucle avec les modèles Spip

    11 avril 2018, par b b

    Un truc que je ne comprends pas, c’est que tu signales dans les points 7 & 8 que "Cela fonctionne sur la version minimal mais pas sur la version production." & "Le bug n’apparait plus..". Tu as donc trouvé l’origine du problème ou non ?

  • How to merge audio with image using transloadit api by passing the input files from http url ?

    23 septembre 2016, par maniempire

    I want to merge audio with image. I tried the transloadit api and using their documentation, i used ruby sdk to implement that. But, when i try to send both the input via http url, i am facing the ffmpeg issue as like below.

    ffmpeg version 2.2.3-transloadit-static-v2.2.3 Copyright (c) 2000-2014 the FFmpeg developers
    built on Jun 3 2014 14:36:03 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
    configuration: --disable-devices --disable-doc --disable-ffplay --disable-ffserver --disable-shared --enable-bzlib --enable-gpl --enable-gray --enable-libass --enable-libfaac --enable-libfdk_aac --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads --enable-runtime-cpudetect --enable-static --enable-version3 --enable-zlib --extra-cflags='-I/usr/src/ffmpeg-static/target/include -static' --extra-cflags=--static --extra-ldflags='-L/usr/src/ffmpeg-static/target/lib -lm -lopus -static' --extra-libs='-lfontconfig -lfreetype -lexpat -lpng -lfribidi -xml2' --extra-version=transloadit-static-v2.2.3 --prefix=/usr/src/ffmpeg-static/target
    libavutil 52. 66.100 / 52. 66.100
    libavcodec 55. 52.102 / 55. 52.102
    libavformat 55. 33.100 / 55. 33.100
    libavdevice 55. 10.100 / 55. 10.100
    libavfilter 4. 2.100 / 4. 2.100
    libswscale 2. 5.102 / 2. 5.102
    libswresample 0. 18.100 / 0. 18.100
    libpostproc 52. 3.100 / 52. 3.100
    Input #0, image2, from '/srv/shared/tmp/scratch/c392b5407fec11e68fdddd7bd9e2f0db/image_%05d.jpg':
    Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: mjpeg, yuvj420p(pc), 1680x1050 [SAR 1:1 DAR 8:5], 25 fps, 25 tbr, 25 tbn, 25 tbc
    [mp3 @ 0x3840540] Format mp3 detected only with low score of 24, misdetection possible!
    [mp3 @ 0x3840540] Estimating duration from bitrate, this may be inaccurate
    Input #1, mp3, from '/srv/shared/tmp/download/bd69dd607fec11e6b6f5a5d0952706af_be1eae707fec11e687b473f9b8ddf7ef.mp3':
    Metadata:
    encoded_by : iTunes 10.2.1
    artist : James Delay
    title : Rewinder
    Duration: 00:04:53.94, start: 0.000000, bitrate: 319 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    Stream #1:1: Video: mjpeg, yuvj422p(pc), 3000x2000, 90k tbr, 90k tbn, 90k tbc
    Metadata:
    title :
    comment : Other
    [swscaler @ 0x381c7e0] deprecated pixel format used, make sure you did set range correctly
    [Parsed_pad_1 @ 0x383db60] Negative values are not acceptable.
    [Parsed_scale_0 @ 0x383fbe0] Failed to configure input pad on Parsed_pad_1
    Error opening filters!
    Conversion failed!

    If i upload an audio file, the issue is not there. If i pass the audio file as a http url, i am getting this error.The sample ruby method i used for this conversion is given below.

    def merge_image_with_audio
       # gem install transloadit
       transloadit = Transloadit.new(
         :key    => '....',
         :secret => '....'
       )

       imported_audio = transloadit.step 'imported_audio', '/http/import', :url => "...."

       imported_image = transloadit.step 'imported_image', '/http/import', :url => "..."

       merger = transloadit.step 'merger', '/video/merge',
         :use => {"steps":[{"name":"imported_audio","as":"audio"},{"name":"imported_image","as":"image"}]},
         :result => true,
         :ffmpeg_stack => "v2.2.3",
         :preset => "ipad-high",
         :resize_strategy => "pad"

       assembly = transloadit.assembly(
         :steps => [ imported_audio, imported_image, merger ]
       )

       response = assembly.submit!
       until response.finished?
         sleep 1; response.reload!
       end

       if !response.error?
         puts response
        # handle success
       else
        puts response
       end

     end

    Even though i am able to do the conversion directly via ffmpeg using this thread, i want to separate this process to a 3rd party so that there will not be any load to the server for the conversion process. Any suggestions or solutions in this regard is highly appreciated.
    Thanks.

  • Ffmpeg - Incorrect codec parameters

    29 avril 2015, par DevPM

    I always got the following error, when trying to connect ffmpeg with ffserver :

    But let’s start with the theory :
    I want to Broadcast a stream, which is already existing in the internet (to able to do some analytic stuff on the stream).
    So my idea is as follows :
    i have one ubuntu-server (currently ubuntu-desktop version 12.x on a virtual machine, i will change it to a real server later...).
    and i want to run on the server :
    - ffmpeg, which should record the live-stream (and saves it locally)
    - ffserver, for broadcasting the same live-stream, using the rtp protocoll (because of the timestamp-header in the protocoll).
    so if i understoot the documentation correctly, this is my idea :
    - start ffserver from linux-terminal
    (ffserver -f ~/Desktop/ffserver.conf)
    - start ffmpeg and connect to the feed of ffserver
    (ffmpeg -re -i "http://InternetStreamer.sdp/playlist.m3u8" -vcodec libx264 -s 320x240 -pix_fmt yuv420p -vb 200000 -minrate 200000 -maxrate 200000 -bufsize 2000000 -acodec libmp3lame -ab 128k -ar 44100 -f rtp -an http://localhost:8090/feed1.ffm)
    (I am getting the Stream from the URL "http://InternetStreamer.sdp/playlist.m3u8" and want sent it to ffserver(also i want to save it locally for backup later...))

    now when i start ffmpeg i get the following error :
    Could not write header for output file #0 (incorrect codec parameters ?)

    Last thing to say is that is very important to broadcast the stream with RTP-Protocoll. (Best case would be directly from ffmpeg, because then the RTP-headers are as set as early as possible)

    my server.conf looks like follows :

    Port 8090
    BindAddress 0.0.0.0
    MaxHTTPConnections 2000
    MaxClients 1000
    MaxBandwidth 1000
    CustomLog -
    NoDaemon
    <feed>
    #file ffmpeg http://localhost:8090/feed1.ffm
    #Format rtp
    #File /tmp/feed1.ffm
    #FileMaxSize 200K
    File /tmp/cam1.ffm
    ACL allow 127.0.0.1
    #VideoFrameRate  25
    </feed>
    <stream>
    Feed feed1.ffm
    Format rtp
    #VideoFrameRate  25
    #Hier: alle parameter von ffmpeg angeben!
    #-re -i "http://apasfiisl.apa.at/ipad/orf2_q4a/orf.sdp/playlist.m3u8"
    #-vcodec libx264
    #-s 320x240
    #-pix_fmt yuv420p
    #-vb 200000
    #-minrate 200000
    #-maxrate 200000
    #-bufsize 2000000
    #-acodec libmp3lame
    #-ab 128k
    #-ar 44100
    #AudioBitRate 32
    #AudioChannels 1
    #AudioSampleRate 44100
    #VideoBitRate 64
    #VideoBufferSize 40
    VideoFrameRate 3
    #VideoSize 160x128
    #VideoGopSize 12
    </stream>

    ##################################################################
    # Special streams
    # Server status
    <stream>
    Format status

    # Only allow local people to get the status
    ACL allow localhost
    ACL allow 192.168.0.0 192.168.255.255

    #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
    </stream>

    # Redirect index.html to the appropriate sites
    <redirect>
    URL http://www.ffmpeg.org/
    </redirect>

    ####################################################################
    This is the full console output from ffmpeg :
    ####################################################################

    The ffmpeg program is only provided for script compatibility and will be removed
    in a future release. It has been deprecated in the Libav project to allow for
    incompatible command line syntax improvements in its replacement called avconv
    (see Changelog for details). Please use avconv instead.
    [applehttp @ 0x91fa240] max_analyze_duration reached
    [applehttp @ 0x91fa240] Estimating duration from bitrate, this may be inaccurate

    Seems stream 1 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 25.00 (25/1)
    Input #0, applehttp, from 'http://InternetStreamer.sdp/playlist.m3u8':
     Duration: N/A, start: 61442.038956, bitrate: N/A
       Stream #0.0: Data: [21][0][0][0] / 0x0015
       Metadata:
         variant_bitrate : 831114
       Stream #0.1: Video: h264 (Main), yuvj420p, 640x360 [PAR 1:1 DAR 16:9], 25 tbr, 90k tbn, 180k tbc
       Metadata:
         variant_bitrate : 831114
       Stream #0.2: Audio: aac, 44100 Hz, stereo, s16
       Metadata:
         variant_bitrate : 831114
    [buffer @ 0x9215720] w:640 h:360 pixfmt:yuvj420p
    [scale @ 0x9434300] w:640 h:360 fmt:yuvj420p -> w:320 h:240 fmt:yuv420p flags:0x4
    [libx264 @ 0x923a560] using SAR=4/3
    [libx264 @ 0x923a560] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2
    [libx264 @ 0x923a560] profile Main, level 1.3
    Output #0, rtp, to 'http://localhost:8090/feed1.ffm':
     Metadata:
       encoder         : Lavf53.21.1
       Stream #0.0: Video: libx264, yuv420p, 320x240 [PAR 4:3 DAR 16:9], q=-1--1, 200 kb/s, 90k tbn, 25 tbc
       Metadata:
         variant_bitrate : 831114
    Stream mapping:
     Stream #0.1 -> #0.0
    Could not write header for output file #0 (incorrect codec parameters ?)

    Kind Regards
    DevPM