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Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...) -
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 -
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 (...)
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FFMPEG ALSA xrun crash
13 décembre 2017, par Liam MartensI’m running a YouTube RTMP stream using FFMPEG with x11grab and an alsa loopback device but sometimes after let’s say 20 hours there is an ALSA xrun and then the ffmpeg command crashes, but I’m not sure why or how this happens. (mind you the ffmpeg command does not run continuously it gets restarted automatically every so often, but the xrun makes the command crash causing the stream to go offline sometimes because a crash restart is not fast enough)
I’m using
thread_queue_sizeand I’ve even manually compiled ffmpeg with a higherALSA BUFFER SIZE, but the issue appears to persist still. Besides this I’ve also scoured many posts with people having similar issues but these never really seem to end up resolved.This is the stream command
ffmpeg -loglevel verbose -f alsa -thread_queue_size 12288 -ac 2 -i hw:Loopback,1,0 \
-probesize 10M -f x11grab -field_order tt -thread_queue_size 12288 -video_size 1280x720 -r 30 -i :1.1 \
-c:v libx264 -c:a libmp3lame -shortest -tune fastdecode -tune zerolatency \
-crf 26 -pix_fmt yuv420p -threads 0 -maxrate 2500k -bufsize 2500k -pass 1 -af aresample=async=1 \
-movflags +faststart -flags +global_header -preset ultrafast -r 30 -g 60 -b:v 2000k -b:a 192k -ar 44100 \
-f flv -rtmp_live live rtmp://a.rtmp.youtube.com/live2/{KEY}Log excerpt
ffmpeg version N-89463-gc7a5e80 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
configuration: --prefix=/usr --enable-avresample --enable-avfilter --enable-gpl --enable-libmp3lame --enable-librtmp --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libtheora --enable-postproc --enable-pic --enable-pthreads --enable-shared --disable-stripping --disable-static --enable-vaapi --enable-libopus --enable-libfreetype --enable-libfontconfig --enable-libpulse --disable-debug
libavutil 56. 5.100 / 56. 5.100
libavcodec 58. 6.103 / 58. 6.103
libavformat 58. 3.100 / 58. 3.100
libavdevice 58. 0.100 / 58. 0.100
libavfilter 7. 7.100 / 7. 7.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 0.101 / 5. 0.101
libswresample 3. 0.101 / 3. 0.101
libpostproc 55. 0.100 / 55. 0.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'hw:Loopback,1,0':
Duration: N/A, start: 1513163617.594224, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Input #1, x11grab, from ':1.1':
Duration: N/A, start: 1513163617.632434, bitrate: N/A
Stream #1:0: Video: rawvideo, 1 reference frame (BGR[0] / 0x524742), bgr0(top first), 854x480, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
Parsing...
Parsed protocol: 0
Parsed host : a.rtmp.youtube.com
Parsed app : live2
RTMP_Connect1, ... connected, handshaking
HandShake: Type Answer : 03
HandShake: Server Uptime : 0
HandShake: FMS Version : 4.0.0.1
HandShake: Handshaking finished....
RTMP_Connect1, handshaked
Invoking connect
HandleServerBW: server BW = 2500000
HandleClientBW: client BW = 10000000 2
HandleChangeChunkSize, received: chunk size change to 256
RTMP_ClientPacket, received: invoke 240 bytes
(object begin)
Property:
Property:
Property:
(object begin)
Property: 3,5,3,824>
Property:
Property:
(object end)
Property:
(object begin)
Property:
Property:
Property:
Property:
Property:
(object begin)
Property:
(object end)
(object end)
(object end)
HandleInvoke, server invoking <_result>
HandleInvoke, received result for method call <connect>
Invoking releaseStream
Invoking FCPublish
Invoking createStream
RTMP_ClientPacket, received: invoke 21 bytes
(object begin)
Property:
Property:
Property: NULL
(object end)
HandleInvoke, server invoking <onbwdone>
Invoking _checkbw
RTMP_ClientPacket, received: invoke 29 bytes
(object begin)
Property:
Property:
Property: NULL
Property:
(object end)
HandleInvoke, server invoking <_result>
HandleInvoke, received result for method call <createstream>
Invoking publish
RTMP_ClientPacket, received: invoke 73 bytes
(object begin)
Property:
Property:
Property: NULL
Property:
(object begin)
Property:
Property:
(object end)
(object end)
HandleInvoke, server invoking <onstatus>
HandleInvoke, onStatus: NetStream.Publish.Start
Stream mapping:
Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
Stream #0:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[graph 0 input from stream 1:0 @ 0x5607d087e060] w:854 h:480 pixfmt:bgr0 tb:1/30 fr:30/1 sar:0/1 sws_param:flags=2
[auto_scaler_0 @ 0x5607d087d800] w:iw h:ih flags:'bicubic' interl:0
[format @ 0x5607d087ed40] auto-inserting filter 'auto_scaler_0' between the filter 'Parsed_null_0' and the filter 'format'
[auto_scaler_0 @ 0x5607d087d800] w:854 h:480 fmt:bgr0 sar:0/1 -> w:854 h:480 fmt:yuv420p sar:0/1 flags:0x4
[swscaler @ 0x5607d0880260] Warning: data is not aligned! This can lead to a speed loss
[libx264 @ 0x5607d08684e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x5607d08684e0] profile Constrained Baseline, level 3.1
[libx264 @ 0x5607d08684e0] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=60 keyint_min=6 scenecut=0 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=26.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1500 vbv_bufsize=1500 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=0
[graph_1_in_0_0 @ 0x5607d091c840] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3
[Parsed_aresample_0 @ 0x5607d0916b40] ch:2 chl:stereo fmt:s16 r:48000Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/{KEY}':
Metadata:
encoder : Lavf58.3.100
Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(top coded first (swapped)), 854x480, q=-1--1, 1000 kb/s, 30 fps, 1k tbn, 30 tbc
Metadata:
encoder : Lavc58.6.103 libx264
Side data:
cpb: bitrate max/min/avg: 1500000/0/1000000 buffer size: 1500000 vbv_delay: -1
Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, s16p, delay 1105, 192 kb/s
Metadata:
encoder : Lavc58.6.103 libmp3lame
frame= 29 fps=0.0 q=17.0 size= 146kB time=00:00:00.94 bitrate=1267.3kbits/s speed=1.86x
frame= 44 fps= 44 q=18.0 size= 168kB time=00:00:01.46 bitrate= 942.4kbits/s speed=1.45x
frame= 60 fps= 40 q=16.0 size= 191kB time=00:00:01.96 bitrate= 794.8kbits/s speed= 1.3x
...
frame= 2740 fps= 30 q=17.0 size= 7993kB time=00:01:31.32 bitrate= 717.0kbits/s speed= 1x
frame= 2755 fps= 30 q=18.0 size= 8013kB time=00:01:31.82 bitrate= 714.9kbits/s speed= 1x
[alsa @ 0x5607d084d7e0] ALSA buffer xrun.
</onstatus></createstream></onbwdone></connect> -
Converting mp3 files to ogg vorbis while retaining cover art using FFmpeg
22 juin 2017, par refi64Title says it all. I’ve tried :
ffmpeg -i 01_FFXV_OST.mp3 -c:a libvorbis 01_FFXV_OST.oggbut I get :
ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
libavutil 55. 44.100 / 55. 44.100
libavcodec 57. 75.100 / 57. 75.100
libavformat 57. 63.100 / 57. 63.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 69.100 / 6. 69.100
libavresample 3. 2. 0 / 3. 2. 0
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Input #0, mp3, from '01_FFXV_OST.mp3':
Metadata:
album : FINAL FANTASY XV Original Soundtrack
artist : Yoko Shimomura
album_artist : SQUARE ENIX MUSIC
composer : Yoko Shimomura
disc : 1
genre : Game
title : Somnus (Instrumental Version)
track : 01
date : 2016
Duration: 00:02:29.47, start: 0.023021, bitrate: 306 kb/s
Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 301 kb/s
Metadata:
encoder : LAME3.98r
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 500x500 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
Automatic encoder selection failed for output stream #0:0. Default encoder for format ogg (codec theora) is probably disabled. Please choose an encoder manually.
Error selecting an encoder for stream 0:0Adding on
-c:v copydoes...nothing :ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
libavutil 55. 44.100 / 55. 44.100
libavcodec 57. 75.100 / 57. 75.100
libavformat 57. 63.100 / 57. 63.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 69.100 / 6. 69.100
libavresample 3. 2. 0 / 3. 2. 0
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Input #0, mp3, from '01_FFXV_OST.mp3':
Metadata:
album : FINAL FANTASY XV Original Soundtrack
artist : Yoko Shimomura
album_artist : SQUARE ENIX MUSIC
composer : Yoko Shimomura
disc : 1
genre : Game
title : Somnus (Instrumental Version)
track : 01
date : 2016
Duration: 00:02:29.47, start: 0.023021, bitrate: 306 kb/s
Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 301 kb/s
Metadata:
encoder : LAME3.98r
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 500x500 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
[ogg @ 0x2ac6e00] Unsupported codec id in stream 0
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:0 -> #0:1 (mp3 (native) -> vorbis (libvorbis))
Last message repeated 1 timesI have absolutely no clue what to do next. I’ve also tried adding
-c:v copyto no avail :ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack --enable-nvenc
libavutil 55. 44.100 / 55. 44.100
libavcodec 57. 75.100 / 57. 75.100
libavformat 57. 63.100 / 57. 63.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 69.100 / 6. 69.100
libavresample 3. 2. 0 / 3. 2. 0
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Input #0, mp3, from '01_FFXV_OST.mp3':
Metadata:
album : FINAL FANTASY XV Original Soundtrack
artist : Yoko Shimomura
album_artist : SQUARE ENIX MUSIC
composer : Yoko Shimomura
disc : 1
genre : Game
title : Somnus (Instrumental Version)
track : 01
date : 2016
Duration: 00:02:29.47, start: 0.023021, bitrate: 306 kb/s
Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 301 kb/s
Metadata:
encoder : LAME3.98r
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 500x500 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
[ogg @ 0x2b6ce00] Unsupported codec id in stream 0
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:0 -> #0:1 (mp3 (native) -> vorbis (libvorbis))
Last message repeated 1 timesI get the same results if I change the stream order (by passing
-map 0:0 -map 0:1). I’m not sure what to do next... -
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