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  • MediaSPIP Core : La Configuration

    9 novembre 2010, par

    MediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
    Il fournit également une page supplémentaire qui n’apparait que lorsque certains plugins sont activés permettant de contrôler l’affichage et les fonctionnalités spécifiques (...)

  • MediaSPIP Player : problèmes potentiels

    22 février 2011, par

    Le lecteur ne fonctionne pas sur Internet Explorer
    Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
    Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (8270)

  • Revision 14e24a1297 : vp9 : enable sse4 sad functions sse4 isn't set by configure or used in rtcd, cor

    1er avril 2015, par James Zern

    Changed Paths :
     Modify /vp9/common/vp9_rtcd_defs.pl



    vp9 : enable sse4 sad functions

    sse4 isn’t set by configure or used in rtcd, correct the sad entries to
    use sse4_1 without changing the signatures for now.
    this was done in vp8 post-vp9 branch.

    Change-Id : Ia9f1fff9f2476fdfa53ed022778dd2f708caa271

  • ffserver leave original stream size

    28 novembre 2014, par ihnatkuk

    Hope you guys will help me, because I have got stuck and can’t find solution for this problem by myself.
    I am trying to stream video from webcam to users using ffmpeg+ffserver. But I have faced with a problem :

    ffmpeg gets stream from camera and pushes it to feed of ffserver:
    ffmpeg -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -y -vcodec libvpx http://127.0.0.1:8090/1.ffm

    ffserver stream options :

    <stream>
    Feed 1.ffm
    Format webm
    NoAudio
    #VideoCodec libvpx
    #VideoSize 480x320
    VideoFrameRate 24
    AVOptionVideo flags +global_header
    AVOptionVideo cpu-used 0
    AVOptionVideo qmin 1
    AVOptionVideo qmax 31
    AVOptionVideo quality good
    PreRoll 0
    StartSendOnKey
    VideoBitRate 128
    </stream>

    (note, videoSize option is commented). But even with default VideoSize (160x128), ffserver doesn’t respond for each request. Browser always gets :

    HTTP/1.0 200 OK
    Pragma: no-cache
    Content-Type: video/webm

    But sometimes video content is not sent.

    If I uncomment VideoSize option - the same problem but much less successfull requests comparing with default video size.

    ffserver log looks regular with no errors. But as you can see that sometimes it sends only headers to client :

    Thu Nov 27 12:49:11 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    Thu Nov 27 12:49:25 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    Thu Nov 27 12:49:36 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    Thu Nov 27 12:50:52 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    Thu Nov 27 12:53:54 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    Thu Nov 27 13:30:19 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
    Thu Nov 27 13:30:34 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 385731
    Thu Nov 27 13:30:34 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 458752
    Thu Nov 27 13:30:36 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
    Thu Nov 27 13:30:58 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 493
    Thu Nov 27 13:30:58 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 622592

    Does anybody know what could it be ? Actually I need to save original VideoSize for stream. I am trying to override ffserver stream options with ffmpeg using the command (passing the same parameters as in ffserver’s stream) :

    ffmpeg -re -override_ffserver -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -an -r 24 -qmin 1 -qmax 31 -cpu-used 0 -quality good -flags:v +global_header -b:v 128 -vcodec libvpx -f webm -y http://127.0.0.1:8090/1.ffm

    But at the momment I still have error message ’Output file is empty, nothing was encoded’. Here is ffmpeg’s output :

    ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
     built on Oct  6 2014 17:33:05 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
     configuration: --prefix=/opt/ffmpeg --libdir=/opt/ffmpeg/lib/ --enable-shared --enable-avresample --disable-stripping --enable-gpl --enable-version3 --enable-runtime-cpudetect --build-suffix=.ffmpeg --enable-postproc --enable-x11grab --enable-libcdio --enable-vaapi --enable-vdpau --enable-bzlib --enable-gnutls --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libfaac --enable-libvo-aacenc --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfdk_aac --enable-libopus --enable-pthreads --enable-zlib --enable-libvpx --enable-libfreetype --enable-libpulse --enable-debug=3
     libavutil      54.  7.100 / 54.  7.100
     libavcodec     56.  1.100 / 56.  1.100
     libavformat    56.  4.101 / 56.  4.101
     libavdevice    56.  0.100 / 56.  0.100
     libavfilter     5.  1.100 /  5.  1.100
     libavresample   2.  1.  0 /  2.  1.  0
     libswscale      3.  0.100 /  3.  0.100
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  0.100 / 53.  0.100
    Guessed Channel Layout for  Input Stream #0.1 : mono
    Input #0, rtsp, from 'rtsp://admin:admin@192.168.10.76:80':
     Metadata:
       title           : RTSP Session/2.0
     Duration: N/A, start: 0.000000, bitrate: 128 kb/s
       Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 100 tbr, 90k tbn, 50 tbc
       Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
    [swscaler @ 0x197f7a0] deprecated pixel format used, make sure you did set range correctly
    [libvpx @ 0x1a0c080] Bitrate 128 is extremely low, maybe you mean 128k
    [libvpx @ 0x1a0c080] v1.3.0
    The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
    Output #0, webm, to 'http://127.0.0.1:8090/1.ffm':
     Metadata:
       title           : RTSP Session/2.0
       encoder         : Lavf56.4.101
       Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=1-31, 0 kb/s, 24 fps, 1k tbn, 24 tbc
       Metadata:
         encoder         : Lavc56.1.100 libvpx
    Stream mapping:
     Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
    Press [q] to stop, [?] for help
    frame=   33 fps= 22 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A dup=0 droframe=   43 fps= 22 q=0.0 Lsize=       0kB time=00:00:00.00 bitrate=N/A dup=0 drop=1    
    video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
    Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
    Received signal 2: terminating.

    Thanks in advance.

  • Ffmpeg command to upload output to S3 using AWS CLI [closed]

    10 novembre 2020, par kmrinmoy07

    I have been trying to upload the output of ffmpeg into S3 using ASW CLI as shown in this post, but this works for video only. Can you give me a ffmpeg command which works for audio files for changing its quality.

    &#xA;