Recherche avancée

Médias (0)

Mot : - Tags -/xmlrpc

Aucun média correspondant à vos critères n’est disponible sur le site.

Autres articles (82)

  • 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 (...)

  • Configurer la prise en compte des langues

    15 novembre 2010, par

    Accéder à la configuration et ajouter des langues prises en compte
    Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
    De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
    Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...)

  • Les tâches Cron régulières de la ferme

    1er décembre 2010, par

    La gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
    Le super Cron (gestion_mutu_super_cron)
    Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...)

Sur d’autres sites (9967)

  • How to process remote audio/video stream on WebRTC server in real-time ? [closed]

    7 septembre 2020, par Kartik Rokde

    I'm new to audio/video streaming. I'm using AntMedia Pro for audio/video conferencing. There will be 5-8 hosts who will be speaking and the expected audience size would be 15-20k (need to mention this as it won't be a P2P conferencing, but an MCU architecture).

    


    I want to give a feature where a user can request for "convert voice to female / robot / whatever", which would let the user hear the manipulated voice in the conference.

    


    From what I know is that I want to do a real-time processing on the server to be able to do this. I want to intercept the stream on the server, and do some processing (change the voice) on each of the tracks, and stream it back to the requestor.

    


    The first challenge I'm facing is how to get the stream and/or the individual tracks on the server ?

    


    I did some research on how to process remote WebRTC streams, real-time on the server. I came across some keywords like RTMP ingestion, ffmpeg.

    


    Here are a few questions I went through, but didn't find answers that I'm looking for :

    


      

    1. Receive webRTC video stream using python opencv in real-time
    2. 


    3. Extract frames as images from an RTMP stream in real-time
    4. 


    5. android stream real time video to streaming server
    6. 


    


    I need help in receiving real-time stream on the server (any technology - preferable Python, Golang) and streaming it back.

    


  • ffmpeg - color-grading video material AND display original source as picture-in-picture, using -filter_complex

    5 octobre 2019, par raven

    this is my first post on this forum, so please be gentle in case I accidentally do trip over any forum rules that I would not know of yet :).

    I would like to apply some color-grading to underwater GoPro footage. To quicker gauge the effect of my color settings (trial-and-error, as of yet), would like to see the original input video stream as a PIP (e.g., scaled down to 50% or even 30%), in the bottom-right corner of the converted output movie.

    I have one input movie that is going to be color graded. The PIP should use the original as an input, just a scaled-down version of it.

    I would like to use ffmpeg’s "-filter_complex" option to do the PIP, but all examples I can find on "-filter_complex" would use two already existing movies. Instead, I would like to make the color-corrected stream an on-the-fly input to "-filter_complex", which then renders the PIP.

    Is that doable, all in one go ?

    Both the individual snippets below work fine, I now would like to combine these and skip the creation of an intermediate color-graded TMP output which then gets combined, with the original, in a final PIP creation process.
    Your help combining these two separate steps into one single "-filter_complex" action is greatly appreciated !

    Thanks in advance,
    raven.

    [existing code snippets (M$ batch files)]

    ::declarations/defines::
    set "INPUT="
    set "TMP="
    set "OUTPUT="
    set "FFMPG="
    set "QU=9" :: quality settings

    set "CONV='"0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1
    0 -1 0:0 -1 0 -1 5 -1 0 -1 0'"" :: sharpening convolution filter

    ::color-grading part::
    %FFMPG% -i %INPUT% -vf convolution=%CONV%,colorbalance=rs=%rs%:gs=%gs%:bs=%bs%:rm=%rm%:gm=%gm%:bm=%bm%:rh=%rh%:gh=%gh%:bh=%bh% -q:v %QU% -codec:v mpeg4 %TMP%

    ::PIP part::
    %FFMPG% -i %TMP% -i %INPUT% -filter_complex "[1]scale=iw/3:ih/3
    [pip]; [0][pip] overlay=main_w-overlay_w-10:main_h-overlay_h-10" -q:v
    %QU% -codec:v mpeg4 %OUTPUT%

    [/existing code]
  • Using ffmpeg with Flash Media Server and HDS

    20 avril 2012, par Jonathan

    I want to use ffmpeg to encode and publish a live stream to Flash Media Server. In order to support iOS devices, I need to implement HTTP Live Streaming as well. The video needs to be in H.264 format and the audio should be AAC. I don't have much experience working with ffmpeg, and I'm having a hard time getting this to work. This is the command that I've tried (and some variations as well) :

    ffmpeg.exe -threads 15 -f dshow -i video="USB2.0 UVC WebCam":audio="Microphone (Realtek High Defini" \
         -map_channel 0.1.1 -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 \
         -s vga -vb 100k -f flv "rtmp:///livepkgr/livestream1?adbe-live-event=liveevent" \
         -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s qvga -vb 200k \
         -f flv "rtmp:///livepkgr/livestream2?adbe-live-event=liveevent" \
         -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s vga -vb 350k
         -f flv "rtmp:///livepkgr/livestream3?adbe-live-event=liveevent"

    When I run this, it appears to connect to FMS, but then I get a lot of error messages about dropped frames - I'm not sure if ANY frames get encoded successfully. My CPU usage is very high as well. I get a 404 error from FMS when I enter the URL of the *.m3u8 file for one of the individual streams (the main livestream.m3u8 file is accessible though). I have also tried outputting to a file instead of FMS, with no success. All I get is some very garbled sound and no video.

    Any suggestions for what options/commands I should use to get this working ? Is anyone using ffmpeg with FMS to do HTTP Dynamic Streaming / HLS with MP4 video ? I've been struggling to get HDS/HLS working for some time now, and any help would be much appreciated ! It shouldn't make a difference, but I'm using FMS on Amazon EC2 with their AMI image.

    Thanks !