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Les tâches Cron régulières de la ferme
1er décembre 2010, parLa 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 (...) -
Ajouter notes et légendes aux images
7 février 2011, parPour 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 (...) -
Emballe médias : à quoi cela sert ?
4 février 2011, parCe plugin vise à gérer des sites de mise en ligne de documents de tous types.
Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ;
Sur d’autres sites (3790)
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Idea on how to make video maker website ?
13 octobre 2015, par Gulam GajaniI want to create website that create video and user can share that website in any social site.
(example sites :
- flixpress.com
- studio.stupeflix.com
- www.photodex.com
- www.picovico.com )
this type of website i want to create.After long research and come to conclusion to use imagemagick and ffmpeg for my website.I learn the basic of the ffmpeg and imagemagick.
SO I create code like this
for text overlay in ffmpeg i create code like thisffmpeg -i input.mp4 -vf drawtext="fontfile=/usr/share/fonts/TTF/Vera.ttf: \
text='Stack Overflow': fontcolor=white: fontsize=24: box=1: boxcolor=black: \
x=(w-text_w)/2: y=(h-text_h-line_h)/2" -codec:a copy output.flvand for imagemagick use convert to make pics and make slideshow with ffmpeg code.
shell_exec(" ffmpeg -framerate 1*10 -i %d.png -c:v libx264 -r 30 -pix_fmt yuv420p out1.mp4");
This code works fine.No error, nothing.
But the output video does not have the quality that i wanted.
My question is that for create website like i mentioned above.Am i using the right software (ffmpeg and imagemagick).what software or code this type of website used ? Any idea ? any opensource software or program for this ?
I am just newbie and try to learn the things.
Please provide guidance on this what i am doing wrong.Just give the software name of programe which is used for this type of website. I can go through by myself. Please Give some idea on this. -
passing additional values to s3 event notification for lambda consumption
8 septembre 2017, par user1790300I have to write code in react-native that allows a user to upload videos to amazon s3 to be transcoded for consumption by various devices. For the processing after the upload occurs ; I am reviewing two approaches :
1) I can use Lambda with ffmpeg to handle the transcoding immediately after the uploading occurs (my fear here would be the amount of time required to transcode the videos and the effect on pricing if it takes a considerable amount of time).
2) I can have s3 pass an sns message to a rest api after the created event occurs and the rest api generate a rabbitmq message that will be processed by worker that will perform the transcoding using ffmpeg.
Option 1) seems to be the preferable option based on a completion time perspective. How concerned should I be with using 1) considering how long video transcoding might take as opposed to option 2) ?
Also, regardless, I need a way to pass additional parameters to lambda or along the sns messaging that would allow me to somehow associate the user who uploaded the video with their account. Is there a way to pass additional text-based values to s3 to pass along to lambda or along sns when the upload completes, as a caveat I plan to upload the video directly to s3 using the rest layer(found this answer here : http://docs.aws.amazon.com/AmazonS3/latest/API/RESTObjectPUT.html#RESTObjectPUT-responses-examples) ?
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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
The post AppRTC : Google’s WebRTC test app and its parameters first appeared on ginger’s thoughts.