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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (71)
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Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users. -
MediaSPIP Player : les contrôles
26 mai 2010, parLes contrôles à la souris du lecteur
En plus des actions au click sur les boutons visibles de l’interface du lecteur, il est également possible d’effectuer d’autres actions grâce à la souris : Click : en cliquant sur la vidéo ou sur le logo du son, celui ci se mettra en lecture ou en pause en fonction de son état actuel ; Molette (roulement) : en plaçant la souris sur l’espace utilisé par le média (hover), la molette de la souris n’exerce plus l’effet habituel de scroll de la page, mais diminue ou (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (9590)
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FFmpeg noise reduction for live broadcast
4 décembre 2019, par Vsevolod TrofimovNow we use FFmpeg for live broadcast to youtube with following options :
ffmpeg -i "rtsp ://xxx" -vcodec libx264 -t $duration -pix_fmt + -c:v copy -acodec libmp3lame -af "afftdn=nt=w" -ar 44100 -threads 6 -qscale 3 -b:a 712000 -bufsize 512k -f flvfor webinars in our university. what is the best way (FFmpeg filters/option) of background noise reduction in live broadcast ? We need only voice on output, no music.
Noise sources are : outdoor noise, projectors and computer noise, students, electrical noises in ip camera’s audiocard.
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Your guide to cookies, web analytics, and GDPR compliance
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AppRTC : Google’s WebRTC test app and its parameters
23 juillet 2014, par silviaIf you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.
When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).
We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.
Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.
Here are my favourite parameters :
- hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
- stereo=true : turns on stereo audio
- debug=loopback : connect to yourself (great to check your own firewalls)
- tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)
For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .
This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :
- ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
- ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
- tp=[password] : password for the TURN server
- audio=true&video=false : audio-only call
- audio=false : video-only call
- audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
- audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
- asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
- arc=opus/48000 : preferred audio receive codec is opus at 48kHz
- dtls=false : disable datagram transport layer security
- dscp=true : enable DSCP
- ipv6=true : enable IPv6
AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).
Have fun playing with the main and always up-to-date WebRTC application : AppRTC.
UPDATE 12 May 2014
AppRTC now also supports the following bitrate controls :
- arbr=[bitrate] : set audio receive bitrate
- asbr=[bitrate] : set audio send bitrate
- vsbr=[bitrate] : set video receive bitrate
- vrbr=[bitrate] : set video send bitrate
Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true