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Autres articles (82)
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Selection of projects using MediaSPIP
2 mai 2011, parThe examples below are representative elements of MediaSPIP specific uses for specific projects.
MediaSPIP farm @ Infini
The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...) -
Sélection de projets utilisant MediaSPIP
29 avril 2011, parLes exemples cités ci-dessous sont des éléments représentatifs d’usages spécifiques de MediaSPIP pour certains projets.
Vous pensez avoir un site "remarquable" réalisé avec MediaSPIP ? Faites le nous savoir ici.
Ferme MediaSPIP @ Infini
L’Association Infini développe des activités d’accueil, de point d’accès internet, de formation, de conduite de projets innovants dans le domaine des Technologies de l’Information et de la Communication, et l’hébergement de sites. Elle joue en la matière un rôle unique (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
Sur d’autres sites (7112)
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VideoJS for HLS streaming not working on iOS
13 février 2024, par AamesI have a website where I stream audio using HLS (m3u8 playlist + ts files).
I use VideoJS 8.3.0 to handle the stream in the frontend side.
The streaming works in all devices (PC, Android, Mac) except in iOS where it works intermittently and often it blocks.


Here is my videojs initialization :


player = videojs('stream-id', {
 controls: false, autoplay: true, preload: 'auto',
 html5: {
 vhs: {overrideNative: false},
 nativeAudioTracks: true
 }
 });

...

player.src({ type: 'application/x-mpegURL', src: 'https://.../master.m3u8' });




And these are the headers of the m3u8 file :


#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:10
#EXT-X-PROGRAM-DATE-TIME:2024-02-13T09:00:00Z
#EXT-X-MEDIA-SEQUENCE:0
#EXT-X-DISCONTINUITY
#EXTINF:10.007800,
master0.ts
#EXTINF:7.214989,
master1.ts
....



I use ffmpeg to create the playlist from mp3 files :


ffmpeg -i inputFileName -c:a aac -b:a 128k -ac 2 -f hls -hls_time 10 -preset ultrafast -map 0:a -flags -global_header -hls_list_size 0 -hls_flags append_list outputFileName




Am I doing something wrong ? Any help would be appreciated
Thanks


I tried this solution : Videojs HLS m3u8 files are not playing in iOS devices and safari browser but it does not work for me


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get error ffmpeg when running wdio-video-reporter in jenkins and centos
31 janvier 2024, par Ferawati Hartanti Pratiwii got this error when i run wdio-video-reporter on jenkins and the OS is centos 8




2024-01-31T17:54:56.047Z ERROR @wdio/config:ConfigParser: Failed loading configuration file:



file :///var/data/jenkins-slave/workspace/QE_Playground/web-testing-pr-only/test/support/wdio.conf.chrome.js :
undefined
Could not find ffmpeg executable, tried "/var/data/jenkins-slave/workspace/QE_Playground/web-testing-pr-only/node_modules/@ffmpeg-installer/linux-x64/ffmpeg",
"/var/data/jenkins-slave/workspace/QE_Playground/web-testing-pr-only/node_modules/@ffmpeg-installer/ffmpeg/node_modules/@ffmpeg-installer/linux-x64/ffmpeg"
and
"/var/data/jenkins-slave/workspace/QE_Playground/web-testing-pr-only/node_modules/@ffmpeg-installer/linux-x64/ffmpeg"




this is my jenkins env


environment {
 PATH = "/usr/bin/ffmpeg:$PATH"
 }



this is my wdio config


ffmpegPath: '/usr/bin/ffmpeg',
 capabilities: [
 {
 maxInstances: parseInt(process.env.MAX_INSTANCES, 10) || 3,
 specs: [
 `${process.cwd()}/test/**_${folder}/features/**/*.feature`,
 ],
 browserName: 'chrome',
 'goog:chromeOptions': browserOptions,
 acceptInsecureCerts: true,
 },
 ],



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avformat/rtsp : Send mode=record instead of mode=receive in Transport header
15 janvier 2024, par Paul Orlykavformat/rtsp : Send mode=record instead of mode=receive in Transport header
Fixes server compatibility issues with rtspclientsink GStreamer plugin.
>From specification :
RFC 7826 "Real-Time Streaming Protocol Version 2.0" (https://datatracker.ietf.org/doc/html/rfc7826), section 18.54 :
mode : The mode parameter indicates the methods to be supported for
this session. The currently defined valid value is "PLAY". If
not provided, the default is "PLAY". The "RECORD" value was
defined in RFC 2326 ; in this specification, it is unspecified
but reserved. RECORD and other values may be specified in the
future.
RFC 2326 "Real Time Streaming Protocol (RTSP)" (https://datatracker.ietf.org/doc/html/rfc2326), section 12.39 :
mode :
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.mode=receive was always like this, from the initial commit 'a8ad6ffa rtsp : Add listen mode'.
For comparison, Wowza was used to push RTSP stream to. Both GStreamer and FFmpeg had no issues.
Here is the capture of Wowza responding to SETUP request :
200 OK
CSeq : 3
Server : Wowza Streaming Engine 4.8.26+4 build20231212155517
Cache-Control : no-cache
Expires : Mon, 15 Jan 2024 19:40:31 GMT
Transport : RTP/AVP/UDP ;unicast ;client_port=11640-11641 ;mode=record ;source=172.17.0.2 ;server_port=6976-6977
Date : Mon, 15 Jan 2024 19:40:31 GMT
Session : 1401457689 ;timeout=60Test setup :
Server : ffmpeg -loglevel trace -y -rtsp_flags listen -i rtsp ://0.0.0.0:30800/live.stream t.mp4
FFmpeg client : ffmpeg -re -i "Big Buck Bunny - FULL HD 30FPS.mp4" -c:v libx264 -f rtsp rtsp ://127.0.0.1:30800/live.stream
GStreamer client : gst-launch-1.0 videotestsrc is-live=true pattern=smpte ! queue ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=60/1 ! timeoverlay font-desc="Sans, 84" halignment=center valignment=center ! queue ! videoconvert ! tee name=t t. ! x264enc bitrate=9000 pass=cbr speed-preset=ultrafast byte-stream=false key-int-max=15 threads=1 ! video/x-h264,profile=baseline ! queue ! rsink. audiotestsrc ! voaacenc ! queue ! rsink. t. ! queue ! autovideosink rtspclientsink name=rsink location=rtsp ://localhost:30800/live.streamTest results :
modified FFmpeg client -> stock server : ok
stock FFmpeg client -> modified server : ok
modified FFmpeg client -> modified server : ok
GStreamer client -> modified server : okSigned-off-by : Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>