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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 est la première version de MediaSPIP stable.
    Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

Sur d’autres sites (8703)

  • How can a desktop node.js app play audio that may be controlled by the user like a media player ? [closed]

    19 février, par eedefeed

    I'm building a playlist manager that plays music. How can Node.JS play audio files quickly, reliably and with all the basic level features you would expect from a media player, namely :

    


      

    • play
    • 


    • pause
    • 


    • seek
    • 


    • stop
    • 


    • adjust volume
    • 


    


    I'm targetting windows/linux, but a Windows-only solution but be okay (for now.)

    


    I have tried a number of libraries and methods to play audio but it seems none of them are good enough :

    


      

    • Audic : it's reasonably good, but buggy. The play and pause functions sometimes get switched around. I also recall that there are some issues with uncaught exceptions somewhere in the dependencies that crash the entire app.
    • 


    • OBS : since the app is designed with broadcasting in mind, I've tried to use OBS's API to get it to play media. Unfortunately, it sometimes stops playback during some tracks, which is surprising since its underlying library, FFmpeg, plays them without issue.
    • 


    • node-groove : seems like its underlying library, libgroove, only supports linux. I can't find any builds to download, regardless.
    • 


    


    Attempts to use Speaker (which seems pretty good) have also failed because all the decoders have big issues :

    


      

    • Anything using lame - I want support for all audio, not specific formats.
    • 


    • fluent-FFmpeg - this is a wrapper around FFmpeg's CLI interface. It has no play/pause function, but bonus library fluent-FFmpeg-util adds this feature. Unfortunately, its pause takes about 4 seconds to work, which I'm guessing is to do with a buffer being exhausted. This is just too latent. Seek would also work by stopping the CLI process and reloading the file, which seems massively inefficient.
    • 


    • Node Vlc - promising but ancient library that gives me reams of node-gyp errors on install. Poorly documented and no explanation of what the library to do
    • 


    • VLC Client - this library has uncaught exceptions that crash the app. Wrapping in try/catch doesn't help.
    • 


    • sound play - doesn't support play/pause
    • 


    


    NPM's search function is filled with audio players designed to work in browsers, but I'm not building a web app. I guess it's an option but it seems inelegant to the point of rediculous.

    


    So it seems the best option centres around FFmpeg. FFmpeg has libraries, and I'm aware that node has ways to hook into those libraries via some sort of C or C++ compatibility layer. Unfortunately, official documentation is rather dense. Different unofficial guides seem to be recommending conflicting approaches (and might be dated), and it's difficult to work out whether myriad technologies are working in tandem or are alternatives, renames or replacements : node-gyp, node-api, addons, windows-build-tools, nan, C vs C++, Visual Studio. It's difficult to make any decisions or know where to start.

    


    Perhaps, also, another option is to use a Python library to interact with FFmpeg, since initial searches have indicated this might be possible. I wouldn't know whether this is a good option.

    


    So my question is : what's my best option to play audio ? Is it another NPM module that I'm not aware of ? Is it a compatibility layer with FFmpeg libraries ?

    


  • Use ffmpeg to stream live content to azure media services

    19 mars 2018, par Dadicool

    I’ve been trying to stream content to azure media services using ffmpeg as it’s one of the options described here : http://azure.microsoft.com/blog/2014/09/18/azure-media-services-rtmp-support-and-live-encoders/

    My command is :

    ffmpeg -v verbose -i 300.mp4 -strict -2 -c:a aac -b:a 128k -ar 44100 -r 30 -g 60 -keyint_min 60 -b:v 400000 -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7

    I have made sure that the streaming endpoint has one active streaming unit, I also made sure that the channel is actually Ready and I even get it to start streaming (which makes a PublishURL available).

    When I execute the ffmpeg command to start streaming, I keep getting the following error :

    ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers
     built on Dec 30 2014 11:31:18 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
     configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 --arch=x86_64 --enable-runtime-cpudetect
     libavutil      54. 15.100 / 54. 15.100
     libavcodec     56. 13.100 / 56. 13.100
     libavformat    56. 15.102 / 56. 15.102
     libavdevice    56.  3.100 / 56.  3.100
     libavfilter     5.  2.103 /  5.  2.103
     libswscale      3.  1.101 /  3.  1.101
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  3.100 / 53.  3.100
    Routing option strict to both codec and muxer layer
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] overread end of atom 'colr' by 1 bytes
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] stream 0, timescale not set
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] max_analyze_duration 5000000 reached at 5003637 microseconds
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '300.mp4':
     Metadata:
       major_brand     : mp42
       minor_version   : 0
       compatible_brands: mp42isomavc1
       creation_time   : 2014-01-11 05:39:32
       genre           : Trailer
       artist          : Warner Bros.
       title           : 300: Rise of an Empire - Trailer 2
       encoder         : HandBrake 0.9.9 2013051800
       date            : 2014
     Duration: 00:02:33.24, start: 0.000000, bitrate: 7377 kb/s
       Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 (1920x1088), 7219 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc (default)
       Metadata:
         creation_time   : 2014-01-11 05:39:32
         encoder         : JVT/AVC Coding
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 157 kb/s (default)
       Metadata:
         creation_time   : 2014-01-11 05:39:32
       Stream #0:2: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 101x150 [SAR 72:72 DAR 101:150], 90k tbr, 90k tbn, 90k tbc
    rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7: Input/output error

    The Azure blog post clearly states that this should be possible but I can’t find a working example anywhere.

    Environment :

    • MacOS Maverick
    • FFMPEG installed from official build
    • 300.mp4 : 1080p trailer of the latest 300 movie
  • Use ffmpeg to stream live content to azure media services

    9 juin 2016, par Dadicool

    I’ve been trying to stream content to azure media services using ffmpeg as it’s one of the options described here : http://azure.microsoft.com/blog/2014/09/18/azure-media-services-rtmp-support-and-live-encoders/

    My command is :

    ffmpeg -v verbose -i 300.mp4 -strict -2 -c:a aac -b:a 128k -ar 44100 -r 30 -g 60 -keyint_min 60 -b:v 400000 -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7

    I have made sure that the streaming endpoint has one active streaming unit, I also made sure that the channel is actually Ready and I even get it to start streaming (which makes a PublishURL available).

    When I execute the ffmpeg command to start streaming, I keep getting the following error :

    ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers
     built on Dec 30 2014 11:31:18 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
     configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 --arch=x86_64 --enable-runtime-cpudetect
     libavutil      54. 15.100 / 54. 15.100
     libavcodec     56. 13.100 / 56. 13.100
     libavformat    56. 15.102 / 56. 15.102
     libavdevice    56.  3.100 / 56.  3.100
     libavfilter     5.  2.103 /  5.  2.103
     libswscale      3.  1.101 /  3.  1.101
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  3.100 / 53.  3.100
    Routing option strict to both codec and muxer layer
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] overread end of atom 'colr' by 1 bytes
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] stream 0, timescale not set
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] max_analyze_duration 5000000 reached at 5003637 microseconds
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '300.mp4':
     Metadata:
       major_brand     : mp42
       minor_version   : 0
       compatible_brands: mp42isomavc1
       creation_time   : 2014-01-11 05:39:32
       genre           : Trailer
       artist          : Warner Bros.
       title           : 300: Rise of an Empire - Trailer 2
       encoder         : HandBrake 0.9.9 2013051800
       date            : 2014
     Duration: 00:02:33.24, start: 0.000000, bitrate: 7377 kb/s
       Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 (1920x1088), 7219 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc (default)
       Metadata:
         creation_time   : 2014-01-11 05:39:32
         encoder         : JVT/AVC Coding
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 157 kb/s (default)
       Metadata:
         creation_time   : 2014-01-11 05:39:32
       Stream #0:2: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 101x150 [SAR 72:72 DAR 101:150], 90k tbr, 90k tbn, 90k tbc
    rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7: Input/output error

    The Azure blog post clearly states that this should be possible but I can’t find a working example anywhere.

    Environment :

    • MacOS Maverick
    • FFMPEG installed from official build
    • 300.mp4 : 1080p trailer of the latest 300 movie