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Autres articles (85)

  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • Gestion des droits de création et d’édition des objets

    8 février 2011, par

    Par défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;

  • 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 (...)

Sur d’autres sites (8753)

  • Streaming live video from Windows to Android with FFmpeg

    29 avril 2018, par Alan Daniels

    I am capturing live video on a Windows PC and encoding it with FFmpeg. I can stream the content live to another PC using rtsp://[dest_ip:port]/live.sdp as FFmpeg’s output on the sender and using ffplay -rtsp_flags listen rtsp://[dest_ip:port]/live.sdp on the receiver. However, I have to run FFplay before starting the sender. Also, VLC cannot play the rtsp path :

    main debug: net: connecting to 127.0.0.1 port 5555
    
main error: connection failed: Connection refused by peer
    
access_realrtsp error: cannot connect to 127.0.0.1:5555
    
access_realrtsp debug: could not connect to: 127.0.0.1:5555/live.sdp

    However, FFplay and VLC can play something like ffplay rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov.

    On the Android side, I am using the Media API which can play content from a URI. It works with HTTP and RTSP (as far as I know).

    I’ve looked at FFmpeg’s streaming guide : https://trac.ffmpeg.org/wiki/StreamingGuide, but I am still confused about the difference between my path rtsp://[dest_ip:port]/live.sdp and rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov.

    Do I need a streaming server in order to have my content accessible by URI ? Any recommendations since FFserver is depreciated ?

  • Concat Demuxer in FFMPEG Autogen

    21 juin 2021, par JunaidAmjad

    Is there option available for Concat Demuxer in FFMPEG.Autogen library. I tested it in FFMPEG and it works well through the here

    


  • Avconv : Select german stream not highest quality one

    6 novembre 2017, par mblaettermann

    I am converting some input stream from my DVB S2 Card to RTMP.

    Everything works fine after switching to recent avconv and x264 :)

    The only thing I couldn’t find out is, how do I select the right audio stream ?

    The source sometimes has up to 6 audio tracks. Avconv automatically chooses the one with the highest bitrate. However I want to select the "ger" one :

    Here are the streams of ARTE german/french TV Channel for example :

    Input #0, mpegts, from 'http://192.168.1.50:9981/stream/channelid/1035':
    
  Duration: N/A, start: 19083.694722, bitrate: 15576 kb/s
    
  Program 1
    
    Stream #0.0[0xa8], 127, 1/90000: Video: mpeg2video (Main), yuv420p, 544x576 [PAR 32:17 DAR 16:9], 1/50, 15000 kb/s, 25 fps, 90k tb50 tbc
    
    Stream #0.1[0x70](fre), 204, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 192 kb/s
    
    Stream #0.2[0x71](ger), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
    
    Stream #0.3[0x72](eng), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
    
    Stream #0.4[0x73](qaa), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
    
  No Program
    
    Stream #0.5[0x3b], 126, 1/90000: Audio: mp1, 0 channels, s16p

    libav Docs are really not that helpful. Who does now the right syntax ?

    EDIT : I found the -map option : http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option But it is not possible to map by name ? Only by index ?

    Maybe I need to use avprobe then, to find the corrent stream index for "ger".

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