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Autres articles (39)

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • MediaSPIP Player : problèmes potentiels

    22 février 2011, par

    Le lecteur ne fonctionne pas sur Internet Explorer
    Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
    Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

Sur d’autres sites (7090)

  • How can I mux H.264 RTP output into a container using FFMPEG ?

    23 septembre 2013, par Grad

    I am working on the effects of network losses in video transmission. In order to simulate the network losses I use a simple program which drops random RTP packets from the output of H.264 RTP encoding.

    I use Joint Model (JM) 14.2 in order to encode the video. However, I don't use AnnexB format as my output, instead I choose the output as RTP packets. The JM output is generated as RTP packets with RTP headers and payload as a sequence. After that, some of RTP packets are dropped by using a simple program. Then, I decode the output by using also JM and it's error concealment methods. That gives me a YUV file as output. The format of the output is as follows :

        ----------------------------------------------------------------------
        | RTP Header #1 | RTP Payload #1 | RTP Header #2 | RTP Payload #2 |...
        ----------------------------------------------------------------------

    I want to make a subjective test with these bitstreams and it's very inconvenient to crowdsource this subjective test with GBs of video data. So, I want to mux these bitstreams into a container (i.e. AVI) by using FFMPEG. I have tried to decode these bitstreams with FFMPEG and FFPLAY ; however, both of them didn't work. I also tried the following command and it didn't work, either.

       ffmpeg - f h264 -i  -vcodec copy -r 25 out.avi

    Which format or muxer should I use ? Do I need to convert these files to any other format ?

  • H.264 - Green frames when streaming over TCP using FFmpeg

    29 août 2017, par Naseeb Panghal

    I am using FC23 machine and FFMPEG version is 2.8.10

    Using below command, I can stream flv over tcp successfully and able to receive it successfully at receiver side as well. I play it using vlc player. Both audio and video are played well.

    ./ffmpeg -f x11grab -s 1920x1080 -framerate 15 -i :0.0 -f alsa -ac 2 -i hw:1 -vcodec libx264 -r 30 -pix_fmt yuv420p -tune zerolatency -preset ultrafast -acodec aac -strict -2 -ar 48000 -ab 96k -f flv -metadata streamName=naseeb.sdp tcp ://127.0.0.1:6666

    But in actually i need to do this using an application. So i wrote an application in ’C’ language.
    I have done following things in the application.

    1. Open `AVOutputFormat` using below API
       fmt = av_guess_format("flv", NULL , NULL);
    2. Get `AVFormatContext` using below API
       avformat_alloc_output_context2(&oc, fmt, NULL, NULL);
    3. Then added streams(audio and video) using required APIs
    4. Then open codecs using required APIs
    5. Then set the output using below API
       ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);

    (A) If I need to write flv into a file on system then I provide filename in step 5 as muxer.flv.

    • result : When I play muxer.flv using VLC player both video and audio are played well

    (B) If I need to stream flv over tcp then I provide filename as tcp://127.0.0.1:6666.

    • result : When it is streamed over network then Green frames are shown for video (yet audio works very well). Is there an extra setting required when I need to stream it over network ?

    I have no doubt on Receiver side as it works very well with FFmpeg utility (command mentioned above).
    Currently I have downloaded ffmpeg 2.8.10 source code and looking into ffmpeg.c file for some extra setting. Till now I have not found anything helpful.

    Please suggest something why Green frames shown when flv is sent on network where it works well when dump on system hard disk.

  • lower latency LAN video streaming to Android ? [closed]

    8 janvier 2020, par LargeMoneyBanks

    I am trying to stream desktop video from windows to an Android device on the same network, but seeing huge latency even streaming windows to windows.

    Technically streaming to a unity game using this asset that doesnt support udp but DOES support HTTPS,HTTP,HLS,RTSP,RTMP. https://assetstore.unity.com/packages/tools/video/ump-android-ios-56044

    With some help from a friend I am using ffmpeg for capture, encoding, and streaming with this command :

    .\ffmpeg -threads 8 -f gdigrab -framerate 60 -i desktop -c:v libx264 -refs 3 -g 60 -loglevel debug -pix_fmt yuv420p -slices 8 -an -preset ultrafast -tune zerolatency -f mpegts -listen 1 tcp://0.0.0.0:1234

    Latency is still around 4 seconds even playing the stream on the same PC as hosting, and fiddling with the network cache. Lowering resolution, fps etc doesn’t change latency. This is my first endeavor into optimizing any kind of streaming so I am a little lost. Anyone know what I might be doing wrong or how I can get to at least under 1 second ?

    Thank you !