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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 (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (10738)

  • How to apply a LUT filter with FFmpeg

    15 septembre 2015, par SwordBearer

    I want to apply a LUT-filter on a video,so i decide to use FFMPEG,but i don’t know how to apply this LUT image on the video ,

    This is the LUT image :
    enter image description here

    Can anyone tell me how to do this With FFMPEG ?

    Thanks.

  • FFmpeg end of filter semicolon being ignored when evaluating chained filters

    21 janvier 2020, par Peter Lockhart

    I’m new to FFmepg so please forgive me if the terminology is wrong.

    I would like to add text and audio to an existing video file. I’m chaining together 2 drawTexts, then in a different filter, I want to merge audio tracks of the original source video and some background music.

    ffmpeg -i sourceNoText.mp4 -i backgroundMusic.mp3 -filter_complex "drawtext=enable='between(t,0,3.5)':fontfile=burbank.ttf:text='Your name - PETER':fontsize=90:x=(w-text_w)/2:y=(h-th-(h/10)-20):fontcolor='White', drawtext=enable='between(t,11.5,14.75)':fontfile=burbank.ttf:text='You couldn't have done it without Peter':fontcolor='White':fontsize=90:x=(w-text_w)/2:y=(h-th-(h/10)-20);[0:a][1:a]amerge,pan=stero:c0code>

    My understanding is filters are separated by a semicolon, yet when I try to do the audio merging, that portion of the command is being interpreted in the drawtext filter.

    [drawtext @ 0000020fe395bec0] Cannot find color 'c0code>

    Running without the audio filter works well, so I don’t believe there to be a syntax error. What am I missing ? Specifying an input and output stream for the drawText chain still produces the same problem.

    I’m using ffmpeg version 4.2.1.

    Thanks in advance.

  • ffmpeg issues "501 Not Implemented" while recording an RTSP stream

    28 février 2019, par atsushi

    I have a 4K camera (Sony SNC-VB770) streaming RTSP.

    I’m trying to record the stream into files (each has handy length, say, 1hour)
    using a simple script to repeatedly launch ffmpeg (ver 4.1) :

    while : ; do
     # (set $url and $outfile, and then)
     ffmpeg -rtsp_transport tcp -t 3600 -y -i $url -c copy -map 0:0 -b:v 16000k $outfile
    done

    If I run the script on a local PC directly connected to the camera, it works (longer than a week, at least).
    However, if I do the same on a server machine located in a data center, it fails randomly with no error message.
    Sometimes it runs for a few days, sometimes it dies in one minutes.

    Typical output looks like the following :

    # devname: snc-vb770
    # url: rtsp://10.40.35.90/media/video1
    # vb: 16000k
    # datefmt %d%H
    # addtimestamp 0
    no timestamp
    ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers
     built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-18)
     configuration: --prefix=/usr/local/ffmpeg-4.1 --enable-openssl --enable-gpl --enable-version3 --enable-nonfree --enable-shared --enable-libx264 --enable-libvorbis --enable-filter=drawtext --enable-libfreetype --enable-libfribidi --enable-fontconfig
     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
     libswscale      5.  3.100 /  5.  3.100
     libswresample   3.  3.100 /  3.  3.100
     libpostproc    55.  3.100 / 55.  3.100
    Input #0, rtsp, from 'rtsp://10.40.35.90/media/video1':
     Metadata:
       title           : Sony RTSP Server
     Duration: N/A, start: 0.066667, bitrate: N/A
       Stream #0:0: Video: h264 (High), yuv420p(tv, bt709, progressive), 3840x2160 [SAR 1:1 DAR 16:9], 14.99 fps, 14.99 tbr, 90k tbn, 29.97 tbc
    Output #0, mp4, to './2811.mp4':
     Metadata:
       title           : Sony RTSP Server
       encoder         : Lavf58.20.100
       Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 3840x2160 [SAR 1:1 DAR 16:9], q=2-31, 16000 kb/s, 14.99 fps, 14.99 tbr, 90k tbn, 90k tbc
    Stream mapping:
     Stream #0:0 -> #0:0 (copy)
    Press [q] to stop, [?] for help
    [mp4 @ 0x24e4ec0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
    [mp4 @ 0x24e4ec0] pts has no value
    [mp4 @ 0x24e4ec0] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file.
    frame=   33 fps=0.0 q=-1.0 size=    1792kB time=00:00:02.00 bitrate=7332.9kbits/s speed=3.57x
    ...
    frame=  104 fps=8.4 q=-1.0 Lsize=    6532kB time=00:00:06.74 bitrate=7939.6kbits/s speed=0.548x
    video:6531kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.023506%

    I’ve looked into RTSP packet and found an "RTSP/1.0 501 Not Implemented" is sent from ffmpeg to the camera.
    After that the camera eventually sent back "RTSP/1.0 505 RTSP Version not supported" and then ffmpeg quits shortly.

    The "501" packet seems to be generated by libavformat/rtsp.c:ff_rtsp_read_reply(),
    when ffmpeg receive a malformed RTSP packet with method=(null), status_code=0.
    I don’t know why such packets arrive at random timing and who is wrong (maybe the camera, maybe any of network switches or routers in the middle of the network path from the camera to the server machine).
    But anyway, I don’t want the recording to be stopped
    due to those malformed packets.

    Is there any workaround to make ffmpeg ignore invalid RTSP packets and just continue the recording ?

    Additional information :

    • I’ve tested the recording with both ffmpeg ver4.1 and 2.8.4 and no difference observed.

    • No difference observed at lower resolution nor at lower bitrate.

    • I have 3 cameras from various manufacturers in the same network environment.
      All of the three are working without problem for more than a month.
      Only the Sony SNC-VB770 shows the strange behavior.