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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
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 (...) -
ANNEXE : Les plugins utilisés spécifiquement pour la ferme
5 mars 2010, parLe site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)
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Error while opening encoder for output stream #0:1 Webm VP9 live streaming
29 août 2019, par SalemThe source video I use is H264 m3u8 live stream and this is the command I tried
ffmpeg -re -i "http://sorce.com/live.m3u8" -c:v libvpx-vp9 -s 480x360 -keyint_min 60\
-g 60 -speed 5 -tile-columns 4 -frame-parallel 1 -threads 8 -static-thresh 0 \
-max-intra-rate 300 -deadline realtime -lag-in-frames 0 -error-resilient 1 \
-b:v 300k -c:a libvorbis -b:a 64k -ar 44100 -f webm_chunk -audio_chunk_duration 2000 \
-header "/var/www/example.com/live/glass_360.hdr" -chunk_start_index 1 \
/var/www/example.com/live/glass_360_%d.chkI pickup this code from wkiki.webmproject.org
most of the time I got this error message[libvorbis @ 0x5617bae0c240] encoder setup failed Error initializing
output stream 0:1 -- Error while opening encoder for output stream
#0:1 - maybe incorrect parameters such as bit_rate, rate, width or heightHere is FFMPEG command output
Input #0, mpegts, from 'http://sorce.com/live.m3u8':
Duration: N/A, start: 1369.000978, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, mono, fltp, 127 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> vp9 (libvpx-vp9))
Stream #0:1 -> #0:1 (aac (native) -> vorbis (libvorbis))I already add video map before video and audio coding argue but I got same error
-map 0:v:0 -c:v libvpx-vp9 and -map 0:a:0 -c:a libvorbis
I tried new broadcast command it works first after I disable the audio
VP9_DASH_PARAMS="-tile-columns 4 -frame-parallel 1 -speed 6" && \
ffmpeg -y -re -i http://sorce.com/live.m3u8 -c:v libvpx-vp9 -s 480x360 -b:v 150k \
-keyint_min 150 -g 150 ${VP9_DASH_PARAMS} -an -f webm -dash 1 \
video_1280x720_500k.webm && sleep 1 && ffmpeg -f webm_dash_manifest \
-i video_1280x720_500k.webm -c copy -f webm_dash_manifest - \ adaptation_sets "id=0" manifest.mpdthis command didn’t create
manifest.mpd
it created onlyvideo_1280x720_500k.webm
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Streaming from Icecast to Facebook Live with ffmpeg on Ubuntu 16.04
24 juillet 2017, par MatthieuI have a webradio streamed by Liquidsoap+Icecast on a DigitalOcean droplet (Ubuntu 16.04), and I want to combine this audio stream with a simple jpeg image with ffmpeg, transform it to a video stream and send it to Facebook live.
Facebook Live specifications :
Video Format :
We accept video in maximum 720p (1280 x 720) resolution, at 30 frames
per second. (or 1 key frame every 2 seconds). You must send an I-frame
(keyframe) at least once every two seconds throughout the stream..
Recommended max bit rate is 4000 Kbps. Titles must be less than 255
characters otherwise the stream will fail. The Live API accepts H264
encoded video and AAC encoded audio only.Video Length :
240 minute maximum length, with the exception of continuous live (see
above). 240 minute maximum length for preview streams (either through
Live dialog or publisher tools). After 240 minutes, a new stream key
must be generated.Advanced Settings :
Pixel Aspect Ratio : Square. Frame Types : Progressive Scan. Audio
Sample Rate : 44.1 KHz. Audio Bitrate : 128 Kbps stereo. Bitrate
Encoding : CBR.And the ffmpeg command I tried :
ffmpeg -loop 1 -i radio-background.jpg -thread_queue_size 20480 -i http://localhost:8000/radio -framerate 30 -r 30 -acodec aac -strict -2 -c:v libx264 -strict experimental -b:a 128k -pix_fmt yuvj444p -x264-params keyint=60 -b:v 256k -minrate 128k -maxrate 512k -bufsize 768k -f flv 'rtmp://rtmp-api.facebook.com:80/rtmp/'
This is actually working, as Facebook receives the live video and allows me to publish it. But I can’t figured out why there is a lag almost every 2 or 3 seconds. I asked different people to watch the test video, and everyone gets the same problem : every 2 or 3 seconds the playing "freezes" for half a second and seems to load the video, I even can see the loading icon spinning on the screen.
I tried different combinations of values for the following options : -thread_queue_size / -b:v / -minrate / -maxrate / -bufsize. Nothing seems to produce any change.
Video streaming is new for me, I’m not really confortable with the options listed before, so I think I’m missing something here...
Also, note that the icecast audio stream perfectly works, and according to DigitalOcean graphs, the server is not overloaded. So I think my ffmpeg command is wrong.
What ffmpeg parameters would be working for that case ?
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Streaming from Icecast to Facebook Live with ffmpeg on Ubuntu 16.04
14 janvier 2017, par MatthieuI have a webradio streamed by Liquidsoap+Icecast on a DigitalOcean droplet (Ubuntu 16.04), and I want to combine this audio stream with a simple jpeg image with ffmpeg, transform it to a video stream and send it to Facebook live.
Facebook Live specifications :
Video Format :
We accept video in maximum 720p (1280 x 720) resolution, at 30 frames
per second. (or 1 key frame every 2 seconds). You must send an I-frame
(keyframe) at least once every two seconds throughout the stream..
Recommended max bit rate is 4000 Kbps. Titles must be less than 255
characters otherwise the stream will fail. The Live API accepts H264
encoded video and AAC encoded audio only.Video Length :
240 minute maximum length, with the exception of continuous live (see
above). 240 minute maximum length for preview streams (either through
Live dialog or publisher tools). After 240 minutes, a new stream key
must be generated.Advanced Settings :
Pixel Aspect Ratio : Square. Frame Types : Progressive Scan. Audio
Sample Rate : 44.1 KHz. Audio Bitrate : 128 Kbps stereo. Bitrate
Encoding : CBR.And the ffmpeg command I tried :
ffmpeg -loop 1 -i radio-background.jpg -thread_queue_size 20480 -i http://localhost:8000/radio -framerate 30 -r 30 -acodec aac -strict -2 -c:v libx264 -strict experimental -b:a 128k -pix_fmt yuvj444p -x264-params keyint=60 -b:v 256k -minrate 128k -maxrate 512k -bufsize 768k -f flv 'rtmp://rtmp-api.facebook.com:80/rtmp/'
This is actually working, as Facebook receives the live video and allows me to publish it. But I can’t figured out why there is a lag almost every 2 or 3 seconds. I asked different people to watch the test video, and everyone gets the same problem : every 2 or 3 seconds the playing "freezes" for half a second and seems to load the video, I even can see the loading icon spinning on the screen.
I tried different combinations of values for the following options : -thread_queue_size / -b:v / -minrate / -maxrate / -bufsize. Nothing seems to produce any change.
Video streaming is new for me, I’m not really confortable with the options listed before, so I think I’m missing something here...
Also, note that the icecast audio stream perfectly works, and according to DigitalOcean graphs, the server is not overloaded. So I think my ffmpeg command is wrong.
What ffmpeg parameters would be working for that case ?