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Autres articles (97)
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Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
ANNEXE : Les plugins utilisés spécifiquement pour la ferme
5 mars 2010, parLe site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)
Sur d’autres sites (6699)
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Is there any way to resume live h264 playback (client rejoin) using the h.264 codec ?
22 février 2021, par IceCurrently, I'm having a bit of an issue with the MediaSource API for resuming live h264 playback.
My server-side code will keep the first packet from FFmpeg, then dispatch it to clients. However, this works well but raises an issue.


When I restart the stream and it dispatches to the client(s), it goes as expected and this is the result I expect when a client disconnects then has to reconnect to the stream, is the aforementioned result


However, if I were to rejoin the stream, I get this as the result


Also, FFMpeg data is being sent to the client in the picture above, it's just not rendering it for some reason.


Here's my function for playing audio/video frames that I get from the server.


private _playFrame(type: 0 | 1) {
 const src = type === 0 ? this.audioSource : this.videoSource;
 if (!src || src.updating) return;
 const queue = type === 0 ? this.audioFrameQueue : this.videoFrameQueue;
 src.appendBuffer(queue.shift());
 if (this.video.src && this.video.paused) this.video.play().then(() => null);
}



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How to add silent data when no audio stream is flowing
29 décembre 2020, par EdqeThis is what I'm currently doing : How to insert silence pcm data when there is no data piping in


Since fluent ffmpeg can use ffmpeg flags, is there a way to put silent data when no actual audio stream is received.


Example :

/
represents real audio data

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represents silent

/////////////////-----///////----//////////

Receive audio cut data cut data

From example above, put silent data on cut when no real data is received (I hope that explains a bit clearer)

I found this discussion, does this work for the example above ? If so, how can I implement it ?


Thanks in advance !


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ffmpeg rtsp_transport to rtmp issues
16 mars 2015, par RandomCouchI’m working on a project that requires taking rtsp links from youtube, and using ffmpeg to stream those videos to an rtmp server. My solution works, however it is having some issues.
I’m using these settings :
-max_delay 0 -initial_pause 0 -rtsp_transport udp -i " + inputLink + " -vcodec libx264 -acodec mp3 -ab 48k -bit_rate 450 -r 25 -s 640x480 -f flv " + stream
inputLink is replaced with the rtsp link, and stream is replaced with the rtmp server link
So this works but here are the issues I’m having :
- At the beginning of each video, there is a big lag spike/lots of frames dropped, and then the video resyncs and plays normally
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Some videos would crash ffmpeg, with a "Conversion failed" message and many frames dropped during the conversion/stream.
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At the end of each video it would start lagging/ dropping frame, right near the end of the video, in other words it doesn’t end normally, every video ends by lagging out / dropping frames
I’ve been struggling for a long time just to get this working, and now I finally did, I just need to perfect it by taking care of those two issues, if anyone has useful information about the rtsp_transport protocol and how to make it stream with no issues, I would greatly appreciate it. Thanks !