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  • Encodage et transformation en formats lisibles sur Internet

    10 avril 2011

    MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
    Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
    Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...)

  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

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  • Slow audio-video sync drift when merging wav and mp4 with ffmpeg

    30 mai 2016, par charlie80

    I have an mp4 file with only a single video stream (no audio) and a wav audio file that I would like to add to the video using ffmpeg. The audio and the video have been recorded simultaneously during a conference, the former from a mixer output on a PC and the latter from a digital videocamera.

    I am using this ffmpeg command :

    ffmpeg -i incontro3.mp4 -itsoffset 18.39 -i audio_mix.wav -c:v copy -c:a aac final-video.mp4

    where I’m using the -itsoffset 18.39 option since I know that 18.39s is the video-audio delay.

    The problem I’m experiencing is that in the output file, while the audio is perfectly in sync with the video at the beginning, it slowly drifts out of sync during the movie.

    The output if ffprobe on the video file is :

    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'incontro3.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       encoder         : Lavf57.25.100
     Duration: 00:47:22.56, start: 0.000000, bitrate: 888 kb/s
       Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 886 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
       Metadata:
         handler_name    : VideoHandler

    and the ffprobe output for the audio file is :

    Input #0, wav, from 'audio_mix.wav':
     Metadata:
       track           : 5
       encoder         : Lavf57.25.100
     Duration: 00:46:32.20, bitrate: 1411 kb/s
       Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

    I’m using the latest ffmpeg Zeranoe windows build git-9591ca7 (2016-05-25).

    Thanks in anticipation for any help/ideas !


    UPDATE 1 : It looks like the problem is upstream the video-audio merging, and could be in the concatenation and conversion of the MTS files generated by the video camera into the mp4 video. I will follow up as I make any progress in understanding...


    UPDATE 2 : The problem is not in the initial merging of the MTS files generated by the camera. Or, at least, it occurs identically if I merge them with cat or with ffmpeg -f concat


    UPDATE 3 : Following @Mulvya’s suggestion, I observed that the drift rate is constant (at least as far as I can tell judging by eye). I also tried to superimpose the A/V tracks with another software, and the drift is exactly the same, thereby ruling out ffmpeg as culprit. My (bad) feeling is that the issue could be related to the internal clocks of the digital video camera and the laptop used for audio recording running at slightly different rates (see here the report of an identical issue I just found).

  • Is there a MS-DRM client library for linux ?

    19 juin 2016, par qrtt1

    I have used the libraries (libavformat and libavcodec) to decode some MMS streaming urls. But some of them are protected by DRM. When I try to decode them, the library will warn about it

    In libavformat/asfdec.c :

    if (!s->keylen) {
       if (!guidcmp(&g, &ff_asf_content_encryption)) {
           av_log(s, AV_LOG_WARNING, "DRM protected stream detected, decoding will likely fail!\n");
       } else if (!guidcmp(&g, &ff_asf_ext_content_encryption)) {
           av_log(s, AV_LOG_WARNING, "Ext DRM protected stream detected, decoding will likely fail!\n");
       } else if (!guidcmp(&g, &ff_asf_digital_signature)) {
           av_log(s, AV_LOG_WARNING, "Digital signature detected, decoding will likely fail!\n");
       }
    }

    The variable s is the struct AVFormatContext. My question is where to get the key ? It seems to decode it with a DRM key.


    I look the ASF Spec, and try to patch asfdec.c. Now, I have the acquiring license URL and key id. The problem can rewrite as ’Is there a MS-DRM client library for linux ?’ (old : How to decode a MMS streaming url with DRM by ffmpeg ?)

    Is it possible to use the license url and key id to get content key ?

  • FFMpeg - Issue with slicing QT videos

    24 juin 2016, par gkmspl

    We use ffmpeg in one of our applications to slice videos. While it’s working fine for slicing PAL videos, it is not working for QT videos... Here’s the command we use :

    ffmpeg.exe -i "input.mp4" ss startTime -c copy -to stopTime -y "output.mp4"

    Throws an error - "[mp4 @ 0515c240] Could not find tag for codec pcm_s16le in stream #1, codec not currently supported in container"

    The input videos are created by a solid state digital video recording system which records videos from following Input channels :
    6 channels of videos input (DVI-4No.s, PAL-2No.s)
    2 channels of audio input (Left & Right)
    2 channels of MIL STD 1553B bus data
    2 channels of RS422 data

    What could be the issue & how can it be resolved ?