Recherche avancée

Médias (91)

Autres articles (61)

  • 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 ;

  • 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 (...)

  • 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 (10503)

  • FFmpeg remux rtp to mpegts [on hold]

    16 décembre 2013, par Ardoramor

    I am trying to remux rtp stream into mptegts format. I have an SDP file with the following contents :

    v=0
    o=- 0 0 IN IP4 127.0.0.1
    s=Unnamed
    i=N/A
    c=IN IP4 192.168.17.44
    t=0 0
    a=recvonly
    a=orient:portrait
    m=video 8202 RTP/AVP 96
    a=rtpmap:96 H264/90000
    a=fmtp:96 packetization-mode=1;profile-level-id=428028;sprop-parameter-sets=Z0KAKJWgKA9E,aM48gA==;
    a=control:trackID=1

    I execute the following ffmpeg command :

    ffmpeg -i "test.sdp" -f mpegts -vcodec copy "/tmp/test.ts"

    And I get the following information :

    Input #0, sdp, from 'test.sdp':
     Metadata:
       title           : Unnamed
       comment         : N/A
     Duration: N/A, start: 0.066622, bitrate: N/A
       Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
    [mpegts @ 0x1101d4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
    Output #0, mpegts, to '/tmp/test.ts':
     Metadata:
       title           : Unnamed
       comment         : N/A
       encoder         : Lavf53.4.0
       Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
    Stream mapping:
     Stream #0.0 -> #0.0

    I receive the following error :

    [mpegts @ 0x1c85f960] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
    av_interleaved_write_frame(): Operation not permitted

    So I add the suggested bitstream filter :

    ffmpeg -i "test.sdp" -f mpegts -vbsf h264_mp4toannexb "/tmp/test.ts"

    But the h264 encoding now becomes h262 (mpeg2video) :

    ~$ffprobe /tmp/test.ts
    Input #0, mpegts, from '/tmp/test.ts':
     Duration: 00:00:04.13, start: 1.400000, bitrate: 640 kb/s
     Program 1
       Metadata:
         service_name    : Unnamed
         service_provider: FFmpeg
       Stream #0.0[0x100]: Video: mpeg2video (Main), yuv420p, 640x480 [PAR 1:1 DAR 4:3], 104857 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbc

    Is there any way to keep the h264 codec without re-encoding it ? Doing so becomes very CPU intensive.

    Update

    Hopefully this will clear up the issue and remove the off-topic stamp.

    I'm writing an Android app that is based off of SpyDroids streaming architecture. The app communicates with the server, providing it the SDP. The server spawns an ffmpeg process to remux the incoming video stream into mpegts and broadcasts it on multicast (right now just file).

    SpyDroid performs streaming by sending recorded mp4 file through localsocket, received h264 packets, supposedly (according to code removed mp4 h264 prefix [annexb]), wraps it with rtp headrs and sends it on its way. Thus, the RPT stream I get is clearly not originally generated as such.

    As @Wagner Patriota has mentioned, I should add '-vcodec copy'. I had run the remuxing with it before as well but the error is still present (full output) :

    ~$ffmpeg -i "test.sdp" -f mpegts -vcodec copy -vbsf h264_mp4toannexb "/tmp/test.ts"
    ffmpeg version 0.8.6, Copyright (c) 2000-2011 the FFmpeg developers
     built on Jan 30 2012 17:17:54 with gcc 4.5.2
     configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --disable-avisynth --enable-libdc1394 --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libx264 --enable-libxvid --extra-cflags='-O2 -g -m64 -mtune=generic -fPIC' --disable-stripping --disable-demuxer=v4l --disable-demuxer=v4l2 --disable-indev=v4l --disable-indev=v4l2
     libavutil    51.  9. 1 / 51.  9. 1
     libavcodec   53.  7. 0 / 53.  7. 0
     libavformat  53.  4. 0 / 53.  4. 0
     libavdevice  53.  1. 1 / 53.  1. 1
     libavfilter   2. 23. 0 /  2. 23. 0
     libswscale    2.  0. 0 /  2.  0. 0
     libpostproc  51.  2. 0 / 51.  2. 0
    [h264 @ 0x16b4b1c0] concealing 232 DC, 232 AC, 232 MV errors
    [h264 @ 0x16b4b1c0] concealing 63 DC, 63 AC, 63 MV errors
    [h264 @ 0x16b4b1c0] concealing 25 DC, 25 AC, 25 MV errors
    [h264 @ 0x16b4b1c0] concealing 138 DC, 138 AC, 138 MV errors
    [h264 @ 0x16b4b1c0] concealing 69 DC, 69 AC, 69 MV errors
    [sdp @ 0x16b43400] Estimating duration from bitrate, this may be inaccurate

    Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
    Input #0, sdp, from 'test.sdp':
     Metadata:
       title           : Unnamed
       comment         : N/A
     Duration: N/A, start: 0.033256, bitrate: N/A
       Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
    [mpegts @ 0x16b4a4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
    Output #0, mpegts, to '/tmp/test.ts':
     Metadata:
       title           : Unnamed
       comment         : N/A
       encoder         : Lavf53.4.0
       Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
    Stream mapping:
     Stream #0.0 -> #0.0
    Press [q] to stop, [?] for help
    h264_mp4toannexb failed for stream 0, codec copy: Invalid argument
    [mpegts @ 0x16b4a4c0] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
    av_interleaved_write_frame(): Operation not permitted

    The error reports that the invalid argument has been supplied. Increased loglevel does not give any more information. I know that ffmpeg is sometimes finicky with argument order. However, they seen to be in order of documentation as well as suggested order by @Wagner Patriota.

  • How does record functionality in vlc works [on hold]

    14 décembre 2013, par quartz

    I am curious about how does "record" functionality of vlc works.
    1) Does it work like screen capture software like "recordmydesktop" ?
    or
    2) Does it save in-memory the frames and latter combine it with tools like ffmpeg ?
    or
    3) Does it simply copy the required information from original video to new video file ?

    Or something else ?

    I want to know these informations because I am developing some simulation tool using a rendering engine. I want my render output to be stored directly in a video file rather than displayed on monitor. If I know these informations, I think it will help.

    Just give me hints, I don't need complete answer.
    This question is not too broad, since I asked specifically about record functionality of vlc media player.

  • Identify local (LAN) vs live URL FFmpeg

    11 juin 2014, par Tarun Seera

    Hi I am using FFmpeg in my iOS app for stream local and live URL.
    Is there any way to identify the URL which I am opening is local or live.
    Actually I want to specify the rtsp_transport in avformat_open_input accordingly.
    If I do not specify the transport its by default taking UDP.
    In UPD I am losing packets in live stream.So i want to specify the transport TCP for live and default for local url.