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  • Publier sur MédiaSpip

    13 juin 2013

    Puis-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

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP 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 (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (12151)

  • h264_mp4toannexb : Try to avoid four byte startcodes

    14 décembre 2019, par Andreas Rheinhardt
    h264_mp4toannexb : Try to avoid four byte startcodes

    According to the H.264 specifications, the only NAL units that need to
    
have four byte startcodes in H.264 Annex B format are SPS/PPS units and
    
units that start a new access unit. Before af7e953a, the first of these
    
conditions wasn't upheld as already existing in-band parameter sets
    
would not automatically be written with a four byte startcode, but only
    
when they already were at the beginning of their input packets. But it
    
made four byte startcodes be used too often as every unit that is written
    
together with a parameter set that is inserted from extradata received a
    
four byte startcode although a three byte start code would suffice
    
unless the unit itself were a parameter set.
    

    FATE has been updated to reflect the changes. Although the patch leaves
    
the extradata unchanged, the size of the extradata according to the FATE
    
reports changes. This is due to a quirk in ff_h2645_packet_split which
    
is used by extract_extradata : If the input is Annex B, the first zero of
    
a four byte startcode is considered a part of the last unit (if any).
    

    Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
    
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] libavcodec/h264_mp4toannexb_bsf.c
    • [DH] tests/ref/fate/h264-bsf-mp4toannexb
    • [DH] tests/ref/fate/h264_mp4toannexb_ticket2991
    • [DH] tests/ref/fate/h264_mp4toannexb_ticket5927
    • [DH] tests/ref/fate/h264_mp4toannexb_ticket5927_2
    • [DH] tests/ref/fate/segment-mp4-to-ts

  • Use audio stream overs ssh with ffmpeg on web page

    12 février 2020, par Mauricio

    I need to broadcast the audio stream (WiFi camera at my home) on the website hosted on a VPS.

    Here is the command executed from my home :

    ffmpeg -i "rtsp://172.16.0.201:554/user=admin&amp;password=root&amp;channel=1&amp;stream=1.sdp" -bufsize 24k -vn -acodec copy -f mpegts - | ssh -p 22222 root@mydomain.org 'ffmpeg -i - -vn -acodec libvorbis -ac 1 -ar 8000 -ab 64k -f mpegts "rtsp://127.0.0.1:53350/live.sdp"'

    It seems to work because I get

    ffmpeg version n4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 9.2.0 (GCC)
    
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --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-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
    
  libavutil      56. 31.100 / 56. 31.100
    
  libavcodec     58. 54.100 / 58. 54.100
    
  libavformat    58. 29.100 / 58. 29.100
    
  libavdevice    58.  8.100 / 58.  8.100
    
  libavfilter     7. 57.100 /  7. 57.100
    
  libswscale      5.  5.100 /  5.  5.100
    
  libswresample   3.  5.100 /  3.  5.100
    
  libpostproc    55.  5.100 / 55.  5.100
    
ffmpeg version 4.1.4-1~deb10u1 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8 (Debian 8.3.0-6)
    
  configuration: --prefix=/usr --extra-version='1~deb10u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
    
  libavutil      56. 22.100 / 56. 22.100
    
  libavcodec     58. 35.100 / 58. 35.100
    
  libavformat    58. 20.100 / 58. 20.100
    
  libavdevice    58.  5.100 / 58.  5.100
    
  libavfilter     7. 40.101 /  7. 40.101
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  3.100 /  5.  3.100
    
  libswresample   3.  3.100 /  3.  3.100
    
  libpostproc    55.  3.100 / 55.  3.100
    
Guessed Channel Layout for Input Stream #0.1 : mono
    
Input #0, rtsp, from 'rtsp://172.16.0.201:554/user=admin&amp;password=root&amp;channel=1&amp;stream=1.sdp':
    
  Metadata:
    
    title           : RTSP Session
    
  Duration: N/A, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: h264 (Main), yuv420p(progressive), 640x360, 12 fps, 12 tbr, 90k tbn, 24 tbc
    
    Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
    
[mpegts @ 0x5582b6b14c40] frame size not set
    
Output #0, mpegts, to 'pipe:':
    
  Metadata:
    
    title           : RTSP Session
    
    encoder         : Lavf58.29.100
    
    Stream #0:0: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=     146kB time=00:00:17.06 bitrate=  70.0kbits/s speed=1.16x

    But when I look on the VPS I don’t see the port open with netstat.

    Yet the process seems to be running

    root     21801  0.2  2.1 265916 42188 ?        SLs  15:03   0:00 ffmpeg -i - -vn -acodec libvorbis -ac 1 -ar 8000 -ab 64k -f mpegts rtsp://127.0.0.1:53350/live.sdp

    I thought I would access the feed with a simple html page with this kind of code

    <audio controls="controls">
    
<source type="audio/ogg"></source>
    
<em>Désolé, votre navigateur ne supporte pas l'audio HTML5</em>
    
</audio>

    iptables

    Chain INPUT (policy DROP)
    
....
    
ACCEPT     tcp  --  anywhere             anywhere             tcp dpt:53350
    
ACCEPT     udp  --  anywhere             anywhere             udp dpt:53350
    
....
    
Chain FORWARD (policy DROP)
    
ACCEPT     tcp  --  anywhere             anywhere             tcp dpt:53350
    
ACCEPT     udp  --  anywhere             anywhere             udp dpt:53350
    
....
    
Chain OUTPUT (policy DROP)
    
ACCEPT     tcp  --  anywhere             anywhere             tcp dpt:53350
    
ACCEPT     udp  --  anywhere             anywhere             udp dpt:53350
    
....

  • FFmpeg get buffer from stream

    4 janvier 2020, par nonoom

    I have an express route, the user gives me a youtube url, with ytdl-core, i get the stream,
    and finally pass it to fluent-ffmpeg, and I need to get a buffer from the ffmpeg stream to finally send the picture to another api (i don’t save it)
    My code looks like :

    router.get("/youtubegifupload", (req,res)=>{ //get video youtube + informations --> create gif --> upload aws s3
    

    
    try{
    
    stream = ytdl("https://www.youtube.com/watch?v=xYQfk5275NA"); //i took random url for test purposes
    
    }catch(error){
    
      res.status(500).json({success : false, message: "Error while getting youtube video ! "})
    
    }
    
    buffer= []
    
    modifiedstream = ffmpeg(stream)
    
    .fps(30)
    
    .setStartTime(10)
    
    .duration(20)
    
    .noAudio()
    
    .videoCodec('gif')
    
    .format('gif')
    
    .save('./jpp3.gif')
    
    .on('progress', (prog)=>{
    
      console.log(prog);
    
    })
    
    .on('data',(buffer)=>{
    
    })
    
    .on('error', ((error)=>{
    
      res.status(500).json({success : false, message: "Error while modifying youtube video ! "})
    
    }))
    
    .on('end', ()=>{
    
        console.log("send to imgur")
    
        //request imgur - pass buffer as image
    
    }) 
    
});

    Everything worked well with save(path) at the end of the ffmpeg operations, but it was for test purposes (actually i don’t want to save),
    Now, my goal is to get a buffer, so i tried to add .on(’data’, .. add the buffer to the array, and concat.
    But here is my problem :
    Without save(), it do not even launch progress(), ffmpeg is printing nothing, no end, ...

    How to make it work without save() ? How to get this stream and concat into a buffer ?

    *I tried : *

    -Verified imports, added const ffmpegPath = require(’@ffmpeg-installer/ffmpeg’).path ;

    -https://github.com/fluent-ffmpeg/node-fluent-ffmpeg/issues/747 - added videoCodec

    Thanks for your help

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