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Autres articles (50)

  • Mise à jour de la version 0.1 vers 0.2

    24 juin 2013, par

    Explications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
    Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...)

  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • 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 (...)

Sur d’autres sites (7540)

  • 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.

  • 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 No encoder found for codec id 8

    3 septembre 2018, par raspiboele

    I’m using a Raspberry Pi to restreem a (https) .m3u8 stream, to a local MJPEG-stream with FFMPEG.
    Becaus I have a Nest Cam / Nest Hello and I want to import that stream into my Fibaro Homecenter (Smarthome Basestation), but the Fibaro Homecenter only eats the MJPEG format. So I made a Raspberry Pi with FFMPEG to do the trick.

    This is my /etc/ffserver.conf file :

    HTTPPort 8090
    

    
HTTPBindAddress 0.0.0.0
    

    
MaxHTTPConnections 2000
    

    
MaxClients 1000
    

    
MaxBandwidth 1000
    

    
CustomLog -
    

    
<feed>
    
File /tmp/feed1.ffm
    
FileMaxSize 30M
    
</feed>
    

    
<stream>
    
Feed feed1.ffm
    
Format mpjpeg
    
VideoFrameRate 2
    
VideoIntraOnly
    
NoAudio
    
Strict -1
    
</stream>

    My command :

     ffmpeg -i "https://path-to-stream.com/chucklist.m3u8" http://localhost:8090/feed1.ffm

    Output :

    ffmpeg version N-89723-g2ca65fc7b7 Copyright (c) 2000-2018 the FFmpeg developers
    
  built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
    
  configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-gnutls
    
  libavutil      56.  7.100 / 56.  7.100
    
  libavcodec     58.  9.100 / 58.  9.100
    
  libavformat    58.  3.100 / 58.  3.100
    
  libavdevice    58.  0.100 / 58.  0.100
    
  libavfilter     7. 11.101 /  7. 11.101
    
  libswscale      5.  0.101 /  5.  0.101
    
  libswresample   3.  0.101 /  3.  0.101
    
  libpostproc    55.  0.100 / 55.  0.100
    
[hls,applehttp @ 0x1d16e70] Opening 'https://path-to-stream.com/chucklist.ts' for reading
    
[hls,applehttp @ 0x1d16e70] Opening 'https://path-to-stream.com/chucklist.ts' for reading
    
Input #0, hls,applehttp, from 'https://path-to-stream.com/chucklist.m3u8':
    
  Duration: N/A, start: 38789.189067, bitrate: N/A
    
  Program 0
    
    Metadata:
    
      variant_bitrate : 0
    
    Stream #0:0: Data: timed_id3 (ID3  / 0x20334449)
    
    Metadata:
    
      variant_bitrate : 0
    
    Stream #0:1: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuvj420p(pc, bt709), 1152x864 [SAR 1:1 DAR 4:3], 15 tbr, 90k tbn, 30 tbc
    
    Metadata:
    
      variant_bitrate : 0
    
    Stream #0:2: Audio: aac (LC) ([15][0][0][0] / 0x000F), 16000 Hz, mono, fltp
    
    Metadata:
    
      variant_bitrate : 0
    
[tcp @ 0x27b36f0] Connection to tcp://localhost:8090 failed (Connection refused), trying next address
    
[ffm @ 0x237a480] no encoder found for codec id 8
    
http://localhost:8090/feed1.ffm: Invalid argument

    As you can see, I get an error : no encoder found for codec id 8
    It is frutstrating me, because I’m looking for a solution for three days now. I can’t even find a list of encoders and the coresponding ID’s. Do I have to enable maybe something in the ./configure ?

    My target is to get a stream at : http://localhost:8090/test1.mjpg

    Can anyone help me please ?

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