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Géodiversité
9 septembre 2011, par ,
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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
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Podcasting Legal guide
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Creativecommons informational flyer
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Autres articles (109)
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Le profil des utilisateurs
12 avril 2011, parChaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...) -
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 -
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
Sur d’autres sites (12787)
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Convert a file using FFMPEG and upload to AWS S3 Nodejs
6 mai 2020, par codernoob8Hey everyone so quick question I want to allow a user to upload a WebM file and convert it using FFmpeg to mp4. I am using Nodejs for the backend and already have a route that uploads files to Amazon S3 file storage. But let's say I wanted to send that file and not store it anywhere but convert it to mp4 from the request itself is that possible ? If not is it possible to take an s3 file URL and convert it to mp4 ? Can anybody point me in the right direction as to what is possible and the best way to do this ?



basically all I want to do is



const objectUrl = createObjectURL(Blob);
ffmpeg -i objectURL S3OutputLocation




or



ffmpeg -i myS3InputLocation myS3OutputLocation



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ffmpeg stream chrome kiosk mode ubuntu 16.04 server
15 février 2021, par RaulI have a weird out-of-sync issue while using ffmpeg to stream to youtube live a chrome browser from an ub untu 16.04 server.



Issue : output video streamed to youtube has audio/video out of sync, sometimes with as much as 3s



Current flow :



1) start pulseaudio - we using something like this to start it :



pulseaudio --start -vvv --disallow-exit --log-target=syslog --high-priority --exit-idle-time=-1 --daemonize




2) start Xvfb



Xvfb :0 -ac -screen 0 1920x1080x24




3) start chrome linux in kiosk mode



google-chrome --kiosk --disable-gpu --incognito --no-first-run --disable-java --disable-plugins --disable-translate --disk-cache-size=$((1024 * 1024)) --disk-cache-dir=/tmp/chrome/ --user-data-dir=/tmp/chrome/ --force-device-scale-factor=1 --window-size=1920,1080 --window-position=0,0 LOCATION_URL




4) start ffmpeg



ffmpeg -y \
 -thread_queue_size 8192 -rtbufsize 250M -f x11grab -video_size 1920x1080 -framerate 24 -i :0 \
 -thread_queue_size 8192 -channel_layout stereo -f alsa -i pulse \
 -c:v libx264 -pix_fmt yuv420p -c:v libx264 -g 48 -crf 24 -filter:v fps=24 -preset ultrafast -tune zerolatency \
 -c:a aac -strict -2 -channel_layout stereo -ab 96k -ac 2 -flags +global_header \
 -f flv YOUTUBE_LIVE_STREAMING_RTMP




Note : this is running on an amazon ec2 instance, meaning there is no soundcard, so alsa and pulseaudio are creating a dummy audio card. However, the latency does not come from there. Logs :



Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Final latency 69.66 ms = 23.22 ms + 2*11.61 ms + 23.22 ms




At this point, here's what we observed :



- 

-
if we start ffmpeg exactly after issuing the command to start chrome, we see the DTS errors from ffmpeg. Audio is out of sync with the video and has delay of 3-5seconds AHEAD. We also noticed the out of sync remains the same for the full duration of the stream
-
if we start ffmpeg after around 10seconds, audio and video are almost in sync. We then manually added a -itsoffset -0.125 to the ffmpeg command and everything is perfect.







Questions :



- 

- Why would ffmpeg have so much lag if it's started right after chrome ?
- Is starting the ffmpeg after 10s or X seconds the expected behavior ? That is, is this because the system needs to wait for audio/video signals to be "ready" or something ?
- Is there a way to 100% calculate or know when Chrome is fully ready and start ffmpeg ? We found sometimes it takes 5s, sometimes 10. Depends on the URL we load.
- Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything. And a restart is required to "re-balance" the audio/video inputs and get them back in sync.
- Can pulseaudio be the problem in this scenario ?













Thank you



UPDATE Dec 20



We were able to do some tricks to force chrome to start the audio on page load, and that will force connect to pulseaudio. Doing that, plus adding a 3s delay for ffmpeg to start, there is no more delay when ffmpeg starts.
However, our app is a webRTC app, and we have a STRANGER thing happening : if we start the page with no webcam/audio, once the webcam/audio is enabled, ffmpeg (while showing no errors) has a delay of 2s or so. While keep talking, in about max 30s, that delay is GONE.



So the new questions are :



- 

- Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything.
- What could cause the initial audio/video out of sync issue and then catching up ?






-
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FFMPEG Incredibly Slow On Windows Server 2016
1er août 2019, par Ben GardnerI have a t2.small (2GB RAM, 1 vCPU) Amazon EC2 instance running a process using FFMPEG. It runs just fine ( 30 fps) on my computer (i7, 12GB RAM), but at around 2 fps on the server. Here’s the command :
rescale_command = f'ffmpeg -i {srcVideo} -filter_complex \"scale={owidth}:{oheight}, setsar=1:1, pad={dim[0]}:{dim[1]}:{oofx}:{oofy}\" {destVideo}'
I’ve tried uninstalling and reinstalling ffmpeg/ffprobe
Edited per llogan’s request, I’ve also discovered that it’s relatively speedy towards the beginning (I’m rescaling 100 times over the course of the program) and slows down towards the end.
It utilizes 100% of the CPU even from the beginning, though.
Example command/output :
ffmpeg -i media/8-1-2019/hi/dl-107.mp4 -filter_complex "scale=607:1080, setsar=1:1, pad=1920:1080:656:0" media/8-1-2019/hi/sl-107.mp4
ffmpeg version 3.2 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.100 / 57. 64.100
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'media/8-1-2019/hi/dl-107.mp4':
Metadata:
minor_version : 512
major_brand : isom
compatible_brands: isomiso2avc1mp41
comment : vid:v09044ce0000bks3802jqrog167l3rf0
encoder : Lavf58.20.100
Duration: 00:00:15.12, start: 0.000000, bitrate: 1046 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 540x960 [SAR 1:1 DAR 9:16], 972 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 64 kb/s (default)
Metadata:
handler_name : SoundHandler
[Parsed_setsar_1 @ 00000000023ac840] num:den syntax is deprecated, please use num/den or named options instead
[Parsed_setsar_1 @ 00000000023ad2c0] num:den syntax is deprecated, please use num/den or named options instead
[libx264 @ 00000000023ad800] using SAR=1/1
[libx264 @ 00000000023ad800] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 00000000023ad800] profile High, level 4.0
[libx264 @ 00000000023ad800] 264 - core 148 r2721 72d53ab - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'media/8-1-2019/hi/sl-107.mp4':
Metadata:
minor_version : 512
major_brand : isom
compatible_brands: isomiso2avc1mp41
comment : vid:v09044ce0000bks3802jqrog167l3rf0
encoder : Lavf57.56.100
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc (default)
Metadata:
encoder : Lavc57.64.100 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
encoder : Lavc57.64.100 aac
Stream mapping:
Stream #0:0 (h264) -> scale (graph 0)
pad (graph 0) -> Stream #0:0 (libx264)
Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
frame= 453 fps=3.6 q=29.0 Lsize= 3016kB time=00:00:15.01 bitrate=1645.7kbits/s speed=0.12x
video:2763kB audio:237kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.576820%
[libx264 @ 00000000023ad800] frame I:6 Avg QP:20.93 size: 38467
[libx264 @ 00000000023ad800] frame P:118 Avg QP:23.12 size: 11624
[libx264 @ 00000000023ad800] frame B:329 Avg QP:25.19 size: 3725
[libx264 @ 00000000023ad800] consecutive B-frames: 1.8% 2.6% 4.6% 90.9%
[libx264 @ 00000000023ad800] mb I I16..4: 16.4% 76.7% 6.9%
[libx264 @ 00000000023ad800] mb P I16..4: 2.3% 5.8% 0.9% P16..4: 9.8% 3.8% 1.6% 0.0% 0.0% skip:75.7%
[libx264 @ 00000000023ad800] mb B I16..4: 0.4% 0.7% 0.1% B16..8: 9.5% 1.6% 0.3% direct: 0.6% skip:86.8% L0:43.0% L1:50.8% BI: 6.2%
[libx264 @ 00000000023ad800] 8x8 transform intra:67.4% inter:78.3%
[libx264 @ 00000000023ad800] coded y,uvDC,uvAC intra: 32.5% 40.1% 9.4% inter: 2.7% 3.9% 0.1%
[libx264 @ 00000000023ad800] i16 v,h,dc,p: 50% 25% 6% 19%
[libx264 @ 00000000023ad800] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 11% 27% 3% 3% 5% 3% 6% 3%
[libx264 @ 00000000023ad800] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 43% 11% 12% 5% 7% 9% 4% 7% 3%
[libx264 @ 00000000023ad800] i8c dc,h,v,p: 62% 11% 22% 5%
[libx264 @ 00000000023ad800] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 00000000023ad800] ref P L0: 64.4% 14.3% 16.1% 5.2%
[libx264 @ 00000000023ad800] ref B L0: 90.6% 7.5% 2.0%
[libx264 @ 00000000023ad800] ref B L1: 97.7% 2.3%
[libx264 @ 00000000023ad800] kb/s:1496.84
[aac @ 0000000000628120] Qavg: 754.761