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  • Les formats acceptés

    28 janvier 2010, par

    Les commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
    ffmpeg -codecs ffmpeg -formats
    Les format videos acceptés en entrée
    Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
    Les formats vidéos de sortie possibles
    Dans un premier temps on (...)

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

Sur d’autres sites (7649)

  • 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".