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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
Autres articles (44)
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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (8189)
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Using ffmpeg with Flash Media Server and HDS
20 avril 2012, par JonathanI want to use ffmpeg to encode and publish a live stream to Flash Media Server. In order to support iOS devices, I need to implement HTTP Live Streaming as well. The video needs to be in H.264 format and the audio should be AAC. I don't have much experience working with ffmpeg, and I'm having a hard time getting this to work. This is the command that I've tried (and some variations as well) :
ffmpeg.exe -threads 15 -f dshow -i video="USB2.0 UVC WebCam":audio="Microphone (Realtek High Defini" \
-map_channel 0.1.1 -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 \
-s vga -vb 100k -f flv "rtmp:///livepkgr/livestream1?adbe-live-event=liveevent" \
-r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s qvga -vb 200k \
-f flv "rtmp:///livepkgr/livestream2?adbe-live-event=liveevent" \
-r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s vga -vb 350k
-f flv "rtmp:///livepkgr/livestream3?adbe-live-event=liveevent"When I run this, it appears to connect to FMS, but then I get a lot of error messages about dropped frames - I'm not sure if ANY frames get encoded successfully. My CPU usage is very high as well. I get a 404 error from FMS when I enter the URL of the *.m3u8 file for one of the individual streams (the main livestream.m3u8 file is accessible though). I have also tried outputting to a file instead of FMS, with no success. All I get is some very garbled sound and no video.
Any suggestions for what options/commands I should use to get this working ? Is anyone using ffmpeg with FMS to do HTTP Dynamic Streaming / HLS with MP4 video ? I've been struggling to get HDS/HLS working for some time now, and any help would be much appreciated ! It shouldn't make a difference, but I'm using FMS on Amazon EC2 with their AMI image.
Thanks !
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Re-sampling H264 video to reduce frame rate while maintaining high image quality
31 mars 2016, par BrianTheLionHere’s the mplayer output for a video of interest :
br@carina:/tmp$ mplayer foo.mov
mplayer: Symbol `ff_codec_bmp_tags' has different size in shared object, consider re-linking
MPlayer 1.0rc4-4.5.2 (C) 2000-2010 MPlayer Team
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing foo.mov.
libavformat file format detected.
[lavf] stream 0: video (h264), -vid 0
[lavf] stream 1: audio (aac), -aid 0, -alang eng
VIDEO: [H264] 1280x720 24bpp 59.940 fps 2494.2 kbps (304.5 kbyte/s)
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264)
==========================================================================
==========================================================================
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 15999->176400)
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample)
Starting playback...
Movie-Aspect is 1.78:1 - prescaling to correct movie aspect.
VO: [vdpau] 1280x720 => 1280x720 Planar YV12I’d like to use ffmpeg, mencoder, or some other command-line video transcoder to re-sample this video to a lower framerate without loss of image quality. That is, each frame should remain as crisp as possible.
Attempts
ffmpeg -i foo.mov -r 25 -vcodec copy bar.mov
- The target frame rate — 25fps — is achieved but individual frames are "blocky."
mencoder -nosound -ovc copy foo.mov -ofps 25 -o bar.mov
- Videos are effectively un-viewable.
Help !
This seems like a simple enough use case. I’m very surprised that obvious things are not working. Is there something wrong with my approach ?
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FFserver with login & password ?
14 décembre 2013, par ContemplatorI use FFserver with FFmpeg for audio transmission through RTP, and anyone can connect to the stream. How can I add a username and password to make my request like rtmp ://username:password@myserver/... (e.g., VLC) ?
Any guidance would be helpful. Thank you.