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Les Miserables
9 décembre 2019, par
Mis à jour : Décembre 2019
Langue : français
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Somos millones 1
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Pourquoi Obama lit il mes mails ?
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Autres articles (3)
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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 (...) -
Contribute to documentation
13 avril 2011Documentation is vital to the development of improved technical capabilities.
MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
To contribute, register to the project users’ mailing (...) -
Participer à sa documentation
10 avril 2011La documentation est un des travaux les plus importants et les plus contraignants lors de la réalisation d’un outil technique.
Tout apport extérieur à ce sujet est primordial : la critique de l’existant ; la participation à la rédaction d’articles orientés : utilisateur (administrateur de MediaSPIP ou simplement producteur de contenu) ; développeur ; la création de screencasts d’explication ; la traduction de la documentation dans une nouvelle langue ;
Pour ce faire, vous pouvez vous inscrire sur (...)
Sur d’autres sites (3147)
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FFmpeg set_buflen while playing RTMP stream
10 avril 2020, par lorenzopI am using sergey-dryabzhinsky/nginx-rtmp-module to build a RTMP server with nginx.
I am streaming a video with FFmpeg to a nginx RTMP app :



ffmpeg -re -i .\video_glasses.mp4 -f flv rtmp://[domain]/input/cam_1




and i want to process it with Python, so I open the stream launching a FFmpeg process from Python like this :



ffmpeg -i rtmp://nginx/input/cam_1 -map 0:v -f rawvideo -pix_fmt rgb24 -vsync 0 pipe:




Here i use vsync 0 to avoid duplicated frames.
I launch the Python program and it waits for the stream. Once the stream has begun, the program sets up some suff and then begins to process the video. Obiviously some time passes. The program correctly processes about 250 frames of video (about 4 sec @ 30 fps) and then blocks because reading from FFmpeg input blocks. In the meantime, the input streaming process has done streaming about 24sec of video.



My best guess is that the maximum buffer of nginx RTMP is full so everything stops. Default buflen for nginx is 20s. I tried to increase it :



#Sets default buffer length. Usually client sends RTMP set_buflen command before playing and resets this 
#setting. Default is 1000 ms.
buflen 50000s;




but nothing changes. So I think the reading FFmpeg process is resetting buflen. I tried



ffmpeg -bufsize 50M -i rtmp://nginx/input/cam_1 -map 0:v -f rawvideo -pix_fmt rgb24 -vsync 0 pipe:




but like that nothing happens at all and Python script never starts.



Any hints ?
Thanks !!


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Any advice on streaming av1 with gstreamer to mediamtx and webrtcbin ?
8 février 2024, par Israel RobotnickI have gstreamer 1.22.7 with the RS plugin for av1 support.
I'm trying to stream AV1 rtp to mediamtx using gstreamer, but the bigger goal is that my rtspsrc->webrtcbin pipeline will work with av1 as it works with h264\vp8\vp9.


I have gstreamer 1.22.7 with the RS plugin for av1 support.
I've created a few av1 files with ffmpeg using svtav1 and rav1e encoders :


ffmpeg -i h264.mp4 -an -c:v libsvtav1 -preset 5 -crf 30 -g 60 -svtav1-params tune=0:fast-decode=1 -pix_fmt yuv420p test1.mp4

ffmpeg -i h264.mp4 -an -c:v librav1e -preset 5 -crf 30 -g 60 -rav1e-params speed=5:low_latency=true -pix_fmt yuv420p test2.mp4



ffmpeg does not currently support AV1 streaming to rtp\rtsp, so im using gstreamer to do so :


gst-launch-1.0 filesrc location=test1.mp4 ! qtdemux ! av1parse ! rtspclietsink location=rtsp://127.0.0.1:8554/test1



From what I've read, mediaMTX\chrome\VLC in their latest versions support av1 streaming in webrtc\rtsp,
but there are no examples whatsoever on how to do so.


Gstreamer preroll, playing and recording when publishing. Everything seems to be fine. Same in mediamtx logs.


When I try to connect a client to the rtsp path via VLC\FFplay\gstreamer rtspsrc->webrtcbin pipeline I don't
et any image. (though webrtc internals show packets arrive fine, but VLC\ffmpeg cant connect)


Any ideas what can be wrong ? Anyone have experience with encoding+streaming AV1 with gstreamer rtspclientsink ?
If you have any tips on redirecting it to webrtcbin (what I do is rtspsrc...parsebin ! queue ! rtpav1pay ! webrtcbin, which seems to connect to chrome and create the av1 sdp, but there in no image) I would appreciate them as well ( :


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Creating own video server with ffserver/ffmpeg streaming RTSP
24 août 2018, par BrykyzI have created app that plays video when url is provided. It can recieve video via RTSP (already tried to play famous big bunny), but now I want to debug it with my own server. Currently I have downloaded ffserver/ffmpeg and want to create my own video server, but i don’t know how.
I have already tried
ffserver &
sudo ffmpeg -r 24 -i "video.mp4" http://localhost:8090/feed1.ffm
ffplay http://localhost:8090/feed1.ffmbut it doesn’t work. I am trying to achieve same thing that happens when I type to terminal
ffplay rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov
(it should play my video), but I want to use private server.
How can I create server that will stream video via RTSP ?
Thank you for your help !
//Edit :
ffserver.conf
HttpPort 8090
RtspPort 5554
HttpBindAddress 0.0.0.0
MaxClients 1000
MaxBandwidth 10000
NoDaemon
<feed>
File /tmp/feed1.ffm
FileMaxSize 1000M
</feed>
<stream>
Feed feed1.ffm
Format rtp
VideoCodec mpeg4
VideoFrameRate 25
VideoBufferSize 80000
VideoBitRate 100
VideoQMin 1
VideoQMax 5
VideoSize 1920x1080
PreRoll 0
Noaudio
</stream>and I am trying to launch it via
ffserver -d & ffmpeg -re -i "simpsons.mp4" http://localhost:8090/feed1.ffm
and then
ffplay "rtsp://localhost:5554/test.mpeg4"
Output is
[tcp @ 0x7fb0880079c0] Connection to tcp://localhost:5554?timeout=0 failed: Connection refused
rtsp://localhost:5554/test.mpeg4: Connection refused