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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
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 -
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 (7622)
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How can I make ffmpeg open my raw video file ?
29 août 2022, par Mandy WeissI'm trying to convert a raw video-only file to some other formats but all I'm getting ffmpeg to produce is a black video.
There are no error messages displayed and this is what ffprobe shows when analyzing the file :





ffprobe version 3.2.2 Copyright (c) 2007-2016 the FFmpeg developers

 built with Apple LLVM version 8.0.0 (clang-800.0.42.1)

 configuration : —prefix=/usr/local/Cellar/ffmpeg/3.2.2 —enable-shared
 —enable-pthreads —enable-gpl —enable-version3 —enable-hardcoded-tables —enable-avresample —cc=clang —host-cflags= —host-ldflags= —enable-libmp3lame —enable-libx264 —enable-libxvid —enable-opencl —disable-lzma —enable-vda libavutil 55. 34.100 / 55. 34.100 libavcodec 57. 64.101 /
 57. 64.101 libavformat 57. 56.100 / 57. 56.100 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 
 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Input #0, avi, from '1.avi' : Metadata :
 encoder : Lavf54.20.4 Duration : 00:00:57.43, start : 0.000000, bitrate : 83196 kb/s
 Stream #0:0 : Video : rawvideo, pal8, 720x480, 83238 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc




Other input files can be transcoded to a playable outfile file with the same ffmpeg convert command :





ffmpeg -i 1.avi -vcodec h264 output.avi





I know the input video is not corrupted because VLC plays it successfully and can even convert it to other formats, but I want to preserve the original one when trying different configurations using ffmpeg.



Thanks in advance for helping !


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FFMPEG - infinite file video stream to RTMP server
5 août 2022, par samoht9277I'm trying to create an infinite video stream that I can push to a local RTMP server.


I believe it's worth mentioning that I'm not very good with
ffmpeg
, I started using it some weeks ago.

Here is a similat question, but it doesn't fully answer what I need.


What I tried


At my first attempt, I made a bash script that spawned a new
ffmpeg
instance for each video I was trying to stream. I noticed that that the few seconds that takes to spawn the new instance interrupts the stream.

After I noticed this problem, I decided to google some kind of queue system, and I found that
ffmpeg
has a concat function, that reads a file with a custom syntax and queues it.

So I made a script that lists the files in the directory where I store all my videos, modifies the output with
sed
, shuffles it (so the episodes are random), and then it redirects the output to aplaylist.txt


find /mnt/episode/ -print | sed '/**.mp4/!d' | sed "s/^/file '/; s/$/'/" | shuf > playlist.txt



playlist.txt


example :


...
file '/mnt/episode/9/1.mp4'
file '/mnt/episode/8/4.mp4'
file '/mnt/episode/5/11.mp4'
file '/mnt/episode/7/15.mp4'
...



This creates 2 problems :


- 

- For some reason, while using the playlist/queue method, when it finishes a video... it takes like 30 seconds to switch to the next video.




ffmpeg -f concat -safe 0 -i playlist.txt -f v4l2 /dev/video0



(simplified command, using /dev/video0 device for testing purposes)


ffmpeg
does output something after the first video, but i'm not quite sure what it means.

[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55c8abc1b900] st: 1 edit list: 1 Missing key frame while searching for timestamp: 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55c8abc1b900] st: 1 edit list 1 Cannot find an index entry before timestamp: 0.



- 

- Once the playlist finishes, the stream is over. I could run the script that generates the playlist again, but that would mean that the stream cuts, which I don't want.





What I want


I need a way to pick a random file, stream it normally via RTMP and once it finished, automatically pick another one without interrupting the streaming, and keep doing this until I decide to turn it off.


This is the directory format :


episodes
│
├── 1
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 2
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 3
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 4
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 5
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 6
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 7
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
├── 8
│   ├── 1.mp4
│   ├── 2.mp4
│   ├── 3.mp4
│ ...
└── 9
 ├── 1.mp4
 ├── 2.mp4
 ├── 3.mp4
 ...



Gladly appreciate some help here. I tried to make the problem as clear as possible.


Thanks <3


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Change the Sample/Pixel aspect ratio of video with least possible video re-encoding
27 août 2022, par PiTaoI need to have some mp4 videos with square sample (pixel) aspect ratio (1:1) i.e. square pixels. However, as ffprobe shows my video has PAR of 59:54.


I have tried using this command :


ffmpeg -i in.mp4 -c copy -bsf:v "h264_metadata=sample_aspect_ratio=1/1" out.mp4



but still the PAR has not changed in my downstream applicaiton or as I checked it with the ffprobe :


ffprobe version n5.1 Copyright (c) 2007-2022 the FFmpeg developers
 built with gcc 12.1.1 (GCC) 20220730
 configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-shared --enable-version3
 libavutil 57. 28.100 / 57. 28.100
 libavcodec 59. 37.100 / 59. 37.100
 libavformat 59. 27.100 / 59. 27.100
 libavdevice 59. 7.100 / 59. 7.100
 libavfilter 8. 44.100 / 8. 44.100
 libswscale 6. 7.100 / 6. 7.100
 libswresample 4. 7.100 / 4. 7.100
 libpostproc 56. 6.100 / 56. 6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'vimeo_11261121.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf59.27.100
 Duration: 00:00:52.36, start: 0.000000, bitrate: 519 kb/s
 Stream #0:0[0x1](eng): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p(progressive), 720x576 [SAR 1:1 DAR 5:4], 382 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 12800 tbn (default)
 Metadata:
 handler_name : Apple Video Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:1[0x2](eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 130 kb/s (default)
 Metadata:
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]



If I not mistaken, the
ffmpeg
only changes the metadata and not the actual PAR. Is it possible to change the PAR (sample aspect ratio) preferably without much re-encoding ?