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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Automated installation script of MediaSPIP

    25 avril 2011, par

    To overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
    You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
    The documentation of the use of this installation script is available here.
    The code of this (...)

  • Script d’installation automatique de MediaSPIP

    25 avril 2011, par

    Afin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
    Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
    La documentation de l’utilisation du script d’installation (...)

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  • ffmpeg : mix/merge multiple mp3 files, some do not mix

    28 août 2018, par C. Ovidiu

    I am trying to merge multiple mp3 files on top of each other on a CentOS 7 server.

    I am trying with ffmpeg but I have mixed results. When mixing 4 files, the last one for example does not mix with the others and is not audible in the final output.

    If I mix this file with another one or two(so max 3 files merged), it works.

    Is there a limit when merging ? For reference, each file is about 10mb is size and 5:00 minutes long.

    This is the command I am using

    ffmpeg -i /var/www/vhosts/site/httpdocs/uploads/tracks/1.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/2.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/3.mp3 -i /var/www/vhosts/site/httpdocs/uploads/tracks/4.mp3 -filter_complex amerge -ac 2 -c:a libmp3lame -q:a 4 /var/www/vhosts/site/httpdocs/uploads/mix.mp3

    The output after merging is this :

    ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
    
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
    
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
    
  libavutil      54. 31.100 / 54. 31.100
    
  libavcodec     56. 60.100 / 56. 60.100
    
  libavformat    56. 40.101 / 56. 40.101
    
  libavdevice    56.  4.100 / 56.  4.100
    
  libavfilter     5. 40.101 /  5. 40.101
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  2.101 /  1.  2.101
    
  libpostproc    53.  3.100 / 53.  3.100
    
[mp3 @ 0x1c8ba60] Skipping 0 bytes of junk at 1044.
    
Input #0, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/1.mp3':
    
  Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
    Metadata:
    
      encoder         : LAME3.98r
    
    Side data:
    
      replaygain: track gain - 6.000000, track peak - unknown, album gain - unknown, album peak - unknown,
    
[mp3 @ 0x1c8eac0] Skipping 0 bytes of junk at 2446.
    
[mp3 @ 0x1c8eac0] Estimating duration from bitrate, this may be inaccurate
    
Input #1, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/2.mp3':
    
  Metadata:
    
    genre           : Other
    
  Duration: 00:05:44.19, start: 0.000000, bitrate: 320 kb/s
    
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
[mp3 @ 0x1c9d640] Skipping 0 bytes of junk at 1044.
    
Input #2, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/3.mp3':
    
  Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
    
    Stream #2:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
    Metadata:
    
      encoder         : LAME3.98r
    
    Side data:
    
      replaygain: track gain - 3.400000, track peak - unknown, album gain - unknown, album peak - unknown,
    
[mp3 @ 0x1cc2b80] Skipping 0 bytes of junk at 1044.
    
Input #3, mp3, from '/var/www/vhosts/site/httpdocs/uploads/tracks/4.mp3':
    
  Duration: 00:05:44.08, start: 0.025057, bitrate: 320 kb/s
    
    Stream #3:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
    Metadata:
    
      encoder         : LAME3.98r
    
    Side data:
    
      replaygain: track gain - 12.100000, track peak - unknown, album gain - unknown, album peak - unknown,
    
[Parsed_amerge_0 @ 0x1cc34e0] No channel layout for input 1
    
[Parsed_amerge_0 @ 0x1cc34e0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
    
Output #0, mp3, to '/var/www/vhosts/site/httpdocs/uploads/mix.mp3':
    
  Metadata:
    
    TSSE            : Lavf56.40.101
    
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p (default)
    
    Metadata:
    
      encoder         : Lavc56.60.100 libmp3lame
    
Stream mapping:
    
  Stream #0:0 (mp3) -> amerge:in0
    
  Stream #1:0 (mp3) -> amerge:in1
    
  amerge -> Stream #0:0 (libmp3lame)
    
Press [q] to stop, [?] for help
    
size=    2360kB time=00:05:44.03 bitrate=  56.2kbits/s
    
video:0kB audio:2360kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.010468%

    Is there a way to solve this, or at least to know what the issue is ?

    Also, some people recommended sox, but I can’t figure how to install it on CentOS.

    Any other alternatives will also help.

    Thank you

  • FFMPEG failing in AWS Lambda

    18 février 2019, par Zaid Amir

    I am trying to create a transcoding function for short videos. The function is hosted on AWS Lambda. The problem is that AWS lambda seems to be missing something that FFMPEG requires, at least according to Amazon.

    I contacted Amazon earlier and this is their response to the issue :

    We found that the FFMPEG operations require at least libx264 and an
    acc library, both of which will have dependencies of their own. To
    troubleshoot the issue it will involve diving deeper into the full
    dependency chain. We can see that it works in the Amazon Linux
    environment however, the environment is similar but not identical to
    the lambda environment. There can be some dependencies that exist in
    Amazon Linux but not in lambda environment as Lambda runs on the
    container. Here, as FFmpeg is a third party software, diving deeper
    into the dependency chain and verifying the version compatibilities is
    very hard to do. Unfortunately going further, this is bound to go into
    architecture and code support which is out of AWS Support scope 1. I
    hope you understand our limitations. However should FFmpeg support
    have any questions specific to the Lambda platform, please do let us
    know and we will be happy to assist. We will be in better position to
    investigate further once you receive an update from the FFmpeg support
    suggesting an issue from Lambda end.

    Upon AWS suggestion, I contacted FFMPEG on the developers mailing list, my message was rejected with the reason being that its more suited to ffmpeg users mailing list than developers. I sent an email to ’ffmpeg-user@ffmpeg.org’ a week ago and did not get any response yet.

    I then went and built a dynamically linked ffmpeg version making sure to package all libraries, checked ddl on each one, then made a small lambda function that looped over all binaries and ddled each one of them, compared that to the output I got from Amazon Linux and the same dependencies/versions exists on both lambda and the AWS Linux instance yet ffmpeg still fails on lambda.

    You can find a detailed log file here : https://www.datafilehost.com/d/6e5e21bb

    And this is a sample of the errors I’m getting, repeated across the entire log file :

    2018-08-14T12:27:10.874Z [h264 @ 0x65c2fc0] concealing 2628 DC, 2628
    AC, 2628 MV errors in P frame

    2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] channel element 2.11 is not
    allocated

    2018-08-14T12:27:10.874Z Error while decoding stream #0:1 : Invalid
    data found when processing input

    2018-08-14T12:27:10.874Z [h264 @ 0x67e86c0] Invalid NAL unit size
    (108085662 > 1649).

    2018-08-14T12:27:10.874Z [h264 @ 0x67e86c0] Error splitting the input
    into NAL units.

    2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] channel element 2.0 is not
    allocated

    2018-08-14T12:27:10.874Z Error while decoding stream #0:1 : Invalid
    data found when processing input

    2018-08-14T12:27:10.874Z [h264 @ 0x68189c0] Invalid NAL unit size
    (71106974 > 1085).

    2018-08-14T12:27:10.874Z [h264 @ 0x68189c0] Error splitting the input
    into NAL units.

    2018-08-14T12:27:10.874Z [aac @ 0x65d2f00] Pulse tool not allowed in
    eight short sequence.

    This log is generated when trying to perform an HLS transcoding on this file : https://www.datafilehost.com/d/999a4492

    Note that the issue is not related to that file alone nor is it related to HLS, its general and happen on all videos and any ffmpeg command that tries to seek the stream, even tried extracting a single frame from a video using the simplest form possible for example : ffmpeg -ss 00:00:02 -I file.mp4 -vframes 1 -y output.jpg also fails with the same errors in the log file.

    Not sure how to debug this further. Tried enabling debug logs with ‘-loglevel debug’ but did not give me any extra info. Any help or suggestions

  • ffmpeg : video codec ansi not compatible with flv

    11 septembre 2018, par diatomym

    lets say i have 10 video files, encoded with the following command in ffmpeg

    ffmpeg -i input.mp4 -c:v libx264 -preset medium -maxrate 6000k -bufsize 6000k -vf "scale=1280:-1,format=yuv420p" -g 50 -c:a aac -b:a 128k -ac 2 -ar 44100 file.flv

    now that all files match in codec, what i want to do is stream all those files via RTMP. for that, I’ll need to create a concat list. i also want the stream to infinitely repeat those 10 files. to do all these things, I use this command :

    ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i mylist.txt -c copy -f flv rtmp://link.to/RTMP

    when doing that, I get the following error output :

     ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i mylist.txt -c copy -f flv rtmp://link.to/RTMP
    
ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
    
  built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
    
  configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
    
  libavutil      55. 78.100 / 55. 78.100
    
  libavcodec     57.107.100 / 57.107.100
    
  libavformat    57. 83.100 / 57. 83.100
    
  libavdevice    57. 10.100 / 57. 10.100
    
  libavfilter     6.107.100 /  6.107.100
    
  libavresample   3.  7.  0 /  3.  7.  0
    
  libswscale      4.  8.100 /  4.  8.100
    
  libswresample   2.  9.100 /  2.  9.100
    
  libpostproc    54.  7.100 / 54.  7.100
    
Input #0, tty, from 'mylist.txt':
    
  Duration: 00:00:00.24, bitrate: 40 kb/s
    
    Stream #0:0: Video: ansi, pal8, 640x400, 25 fps, 25 tbr, 25 tbn, 25 tbc
    
[flv @ 0x560ec7662920] ***Video codec ansi not compatible with flv
    
Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented***
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)

    ffmpeg is giving me the error

    Video codec ansi not compatible with flv
    Could not write header for output file #0 (incorrect codec parameters ?): Function not implemented

    what am i doing wrong ? thanks for the help.

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