Recherche avancée

Médias (29)

Mot : - Tags -/Musique

Autres articles (65)

  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Other interesting software

    13 avril 2011, par

    We don’t claim to be the only ones doing what we do ... and especially not to assert claims to be the best either ... What we do, we just try to do it well and getting better ...
    The following list represents softwares that tend to be more or less as MediaSPIP or that MediaSPIP tries more or less to do the same, whatever ...
    We don’t know them, we didn’t try them, but you can take a peek.
    Videopress
    Website : http://videopress.com/
    License : GNU/GPL v2
    Source code : (...)

  • 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 (6494)

  • 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".

Navigation