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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (53)
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Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (6149)
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FFMPEG stream from sdp to youtube/facebook live(RTMP)
3 avril 2020, par AkilSimilar to this question, I'm trying to stream to Youtube/Facebook live (rtmp) via Ffmpeg.



The server on which ffmpeg is installed receives a rtp stream (source of the stream uses Vp8 and opus as the codecs), i then use the sdp given below as input for the ffmpeg command



SDP :



v=0
o=- 0 0 IN IP4 127.0.0.1
s=FFMPEG Test
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 56.15.102
m=audio 10004 RTP/AVP 111
a=rtpmap:111 OPUS/48000/2
m=video 10006 RTP/AVP 100
a=rtpmap:100 VP8/90000
a=fmtp:100 packetization-mode=1




Where 10004 and 10006 are the audio and video port of the rtp input stream.



I'm using the following FFMPEG command to stream to youtube/facebook live



ffmpeg -protocol_whitelist file,crypto,udp,rtp -i input.sdp -c:a aac -ar 44100 -b:a 128k -pix_fmt yuv420p -profile:v baseline -s 426x240 -bufsize 6000k -vb 400k -maxrate 1500k -deinterlace -vcodec libx264 -preset veryfast -g 30 -r 30 -f flv "rtmp_url"




Log :



ffmpeg version 4.2.2-1build1~18.04.sav0 Copyright (c) 2000-2019 the FFmpeg developers
 built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
 configuration: --prefix=/usr --extra-version='1build1~18.04.sav0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 31.100 / 56. 31.100
 libavcodec 58. 54.100 / 58. 54.100
 libavformat 58. 29.100 / 58. 29.100
 libavdevice 58. 8.100 / 58. 8.100
 libavfilter 7. 57.100 / 7. 57.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 5.100 / 5. 5.100
 libswresample 3. 5.100 / 3. 5.100
 libpostproc 55. 5.100 / 55. 5.100
 [sdp @ 0x56083912d340] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize' options
 Input #0, sdp, from 'input.sdp':
 Metadata:
 title : FFMPEG Test
 Duration: N/A, start: 0.000000, bitrate: N/A
 Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
 Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
 Codec AVOption profile () specified for output file #0 (youtube_rtmp_url) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
 Codec AVOption g (set the group of picture (GOP) size) specified for output file #0 (youtube_rtmp_url) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
 Codec AVOption preset (Configuration preset) specified for output file #0 (youtube_rtmp_url) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
 Stream mapping:
 Stream #0:0 -> #0:0 (opus (native) -> aac (native))
 Press [q] to stop, [?] for help
 Output #0, flv, to 'youtube_rtmp_url':
 Metadata:
 title : FFMPEG Test
 encoder : Lavf58.29.100
 Stream #0:0: Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 128 kb/s
 Metadata:
 encoder : Lavc58.54.100 aac
 av_interleaved_write_frame(): Broken pipe 134.5kbits/s speed=2.35x 
 [flv @ 0x5608391a7040] Failed to update header with correct duration.
 [flv @ 0x5608391a7040] Failed to update header with correct filesize.
 Error writing trailer of youtube_rtmp_url: Broken pipe
 size= 162kB time=00:00:09.86 bitrate= 134.2kbits/s speed=2.32x 
 video:0kB audio:155kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 4.493091%
 [aac @ 0x5608391a7ac0] Qavg: 247.499
 Conversion failed!




I have verified that ffmpeg can read the stream, i used a different ffmpeg command to record the stream and that works well. Could someone please point out whats wrong or link to some documentation that might help ?


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Streaming to Facebook using ffmpeg
4 octobre 2020, par user7389496I want to stream to Facebook using
ffmpeg
, I'm runningOBS
withMonaServer

This is the command I'm using to start my stream :

ffmpeg -i rtmp://localhost:1935/live/stream -c:v copy -c:a copy -map 0 -f tee "[f=flv]rtmps://live-api-s.facebook.com:443/rtmp/Here-goes-my-stream-key"



but I receive the following error


[flv @ 000001f7a1367740] Failed to update header with correct duration.0.168x
 [flv @ 000001f7a1367740] Failed to update header with correct filesize.
 [tee @ 000001f7a3550740] All tee outputs failed.
 av_interleaved_write_frame(): End of file
 frame= 546 fps=4.5 q=-1.0 Lsize=N/A time=00:00:20.80 bitrate=N/A speed=0.173x
 video:22346kB audio:405kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
 Conversion failed!



Nothing shows on the Facebook live except. Is there a way to solve this ?


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Streaming to Facebook Live from ffmpeg gets terminated
24 avril 2021, par RaghavendraI am trying to capture the canvas on browser (canvas.captureStream), add audio ( captured using getUserMedia and added to stream using canvasStream.addTracks) and send it to server. Server sends the stream ( after encoding with H264 ) to facebook live using ffmpeg. However the stream is not stable and gets disconnected within minutes.
If the video along with audio tracks (using getMediaUser) is directly(without canvas.captureStream) sent it works fine. I am suspecting it has to do with the audio track not being proper and hence facebook rejecting it ( no particular error from ffmpeg, it just exits with IO error). Need help in figuring out the right way to send canvas stream along with audio.


Current ffmpeg command is as below


ffmpeg -i - -c:v libx264 -crf 23 -preset ultrafast \
 -tune zerolatency -max_muxing_queue_size 1000 \
 -vsync cfr -async 1 -bufsize 2M -r 30 -g 60 -keyint_min 30 \
 -x264opts keyint=30 -pix_fmt yuv420p -level 3 \
 -c:a aac -b:a 96k -ar 96000 \
 -f tee -map 0:v -map 0:a [f=flv] <rtmp url="url" 1="1"> | [f=flv] <rtmp url="url" 2="2">
</rtmp></rtmp>


Note : In the above command
and are replaced with actual urls ( including the session key required ) 

My settings to capture canvas and audio on browser are as below



 canvasStream = canvas.captureStream(30);
 audioTrack = stream.getTracks().filter( (track) => {
 return track.kind === 'audio';
 })[0];
 canvasStream.addTrack(audioTrack);




Note : Audio constraints given in getUserMedia are as below


audio: {
 sampleRate: 44100, 
 echoCancellation: true
 }



Error given by ffmpeg while exiting is as below


FFMPEG:[flv @ 0x5626ecf0f7c0] Failed to update header with correct duration.
[flv @ 0x5626ecf0f7c0] Failed to update header with correct filesize.

FFMPEG:[tee @ 0x5626ec8550c0] Slave muxer #1 failed, aborting.
av_interleaved_write_frame(): Broken pipe
FFMPEG:[flv @ 0x5626ecbe8840] Failed to update header with correct duration.

FFMPEG:[flv @ 0x5626ecbe8840] Failed to update header with correct filesize.

FFMPEG:frame= 2378 fps= 22 q=22.0 Lsize=N/A time=00:01:21.17 bitrate=N/A dup=2202 drop=5 speed=0.745x 
video:18916kB audio:959kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown

FFMPEG:[libx264 @ 0x5626ec855b80] frame I:80 Avg QP:13.97 size:115968
[libx264 @ 0x5626ec855b80] frame P:2298 Avg QP:16.77 size: 4392
[libx264 @ 0x5626ec855b80] mb I I16..4: 100.0% 0.0% 0.0%

FFMPEG:[libx264 @ 0x5626ec855b80] mb P I16..4: 3.8% 0.0% 0.0% P16..4: 9.5% 0.0% 0.0% 0.0% 0.0% skip:86.7%
[libx264 @ 0x5626ec855b80] coded y,uvDC,uvAC intra: 26.3% 42.2% 13.6% inter: 2.9% 6.1% 0.1%
[libx264 @ 0x5626ec855b80] i16 v,h,dc,p: 34% 26% 23% 17%
[libx264 @ 0x5626ec855b80] i8c dc,h,v,p: 46% 27% 20% 7%
[libx264 @ 0x5626ec855b80] kb/s:1954.96
[aac @ 0x5626ec84d200] Qavg: 122.161
Conversion failed!



Appreciate any help with this.