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  • 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 (...)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • Ecrire une actualité

    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

Sur d’autres sites (12475)

  • Duration of source video and subtracted audio are different

    22 avril 2018, par zpc

    The duration of source video and subtracted wav audio is different , why ?

    I’m recorgnizing subtitle from audio, and I need to add subtitle back to video. So I want the duration of audio and video the same.

    ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y

    My CLI :

    [zhangpengcheng@mobiledev03v ifly]$ ffprobe http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 2>&1 | grep Duration
     Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s


    [zhangpengcheng@mobiledev03v ifly]$ ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y
    ffmpeg version 3.1.3 Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
     configuration: --prefix=./build/ --enable-shared --enable-static --enable-libx264 --enable-avisynth --enable-libass --enable-libfdk-aac --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libopencv --enable-librtmp --enable-gpl --enable-nonfree
     libavutil      55. 28.100 / 55. 28.100
     libavcodec     57. 48.101 / 57. 48.101
     libavformat    57. 41.100 / 57. 41.100
     libavdevice    57.  0.101 / 57.  0.101
     libavfilter     6. 47.100 /  6. 47.100
     libswscale      4.  1.100 /  4.  1.100
     libswresample   2.  1.100 /  2.  1.100
     libpostproc    54.  0.100 / 54.  0.100
    Input #0, hls,applehttp, from 'http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8':
     Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s
     Program 0
       Metadata:
         variant_bitrate : 0
       Stream #0:0: Video: h264 (Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m/bt709/bt709), 668x376, 15 tbr, 90k tbn, 180k tbc
       Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, mono, fltp, 65 kb/s
    [wav @ 0x10182e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
    Output #0, wav, to 'test.wav':
     Metadata:
       ISFT            : Lavf57.41.100
       Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
       Metadata:
         encoder         : Lavc57.48.101 pcm_s16le
    Stream mapping:
     Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
    Press [q] to stop, [?] for help
    size=   31904kB time=00:06:51.84 bitrate= 634.6kbits/s speed= 146x    
    video:0kB audio:31904kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000239%


    [zhangpengcheng@mobiledev03v ifly]$ ffprobe test.wav 2>&1 | grep Duration
     Duration: 00:06:10.40, bitrate: 705 kb/s
  • FFMPEG HLS stream for Android and IOS

    30 juin, par Poda

    I'm trying to stream to mobile devices with ffmpeg and apache2.2 but I haven't been successful.

    



    I used this command to create the segments and the playlist :

    



    ffmpeg -i http://x.x.x.x:8080 -codec:v libx264 -r 25 -pix_fmt yuv420p -profile:v baseline -level 3 -b:v 500k -s 640x480 -codec:a aac -strict experimental -ac 2 -b:a 128k -movflags faststart -flags -global_header -map 0 -f hls  -hls_time 10 -hls_list_size 5 -hls_allow_cache 0 -sc_threshold 0 -hls_flags delete_segments -hls_segment_filename out%05d.ts list.m3u8


    



    The source is a http stream which is streamed by VLC media player.

    



    Example content of the list.m3u8 file :

    



    #EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:10
#EXT-X-MEDIA-SEQUENCE:89
#EXTINF:10.000000,
out00089.ts
#EXTINF:10.000000,
out00090.ts
#EXTINF:10.000000,
out00091.ts
#EXTINF:10.000000,
out00092.ts
#EXTINF:9.000000,
out00093.ts
#EXT-X-ENDLIST


    



    I created another playlist file - playlist.m3u8 :

    



    #EXTM3U
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=512000
http://x.x.x.x/list.m3u8


    



    If I open this (playlist.m3u8) file in VLC media player then it plays.
It also works in desktop chrome and desktop firefox browsers with Video-js plugin flash fallback.

    



    I set the correct MIME types to the .ts and .m3u8 files in .htaccess file :

    



    AddType application/x-mpegURL .m3u8
AddType video/MP2T .ts


    



    FFprobe output for playlist.m3u8 :

    



    Input #0, hls,applehttp, from 'playlist.m3u8':
    Duration: N/A, start: 1.400000, bitrate: N/A
    Program 0
    Metadata: variant_bitrate : 512000
Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc
Metadata: variant_bitrate : 512000
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 128 kb/s
Metadata: variant_bitrate : 512000


    



    What should I do to make it work ?

    



    UPDATE

    



    It works if I provide a link to list.m3u8 file (created by ffmpeg).

    


  • 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