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Autres articles (77)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (9708)

  • ffmpeg youtube live stream settings screen output [closed]

    25 juin 2020, par brainoverflowse
    

    

    Below works but I do get some warnings:

    ffmpeg -rtsp_transport tcp -i rtsp://admin:password@192.168.0.121:5542/Streaming/Channels/1 -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/streamcode
ffmpeg version git-2020-06-20-29ea4e1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 55.100 / 56. 55.100
  libavcodec     58. 93.100 / 58. 93.100
  libavformat    58. 47.100 / 58. 47.100
  libavdevice    58. 11.100 / 58. 11.100
  libavfilter     7. 86.100 /  7. 86.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://admin:password@192.168.0.121:5542/Streaming/Channels/1':
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 20 fps, 30 tbr, 90k tbn, 40 tbc
    Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
Multiple -c, -codec, -acodec, -vcodec, -scodec or -dcodec options specified for stream 0, only the last option '-c:v copy' will be used.
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_mulaw (native) -> aac (native))
Press [q] to stop, [?] for help
[aac @ 000002a2862cb900] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/streamcode':
  Metadata:
    title           : Media Presentation
    encoder         : Lavf58.47.100
    Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuv420p(progressive), 1920x1080, q=2-31, 20 fps, 30 tbr, 1k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s
    Metadata:
      encoder         : Lavc58.93.100 aac
[flv @ 000002a285413200] Non-monotonous DTS in output stream 0:0; previous: 283, current: 33; changing to 283. This may result in incorrect timestamps in the output file.
[flv @ 000002a285413200] Non-monotonous DTS in output stream 0:0; previous: 283, current: 100; changing to 283. This may result in incorrect timestamps in the output file.
[flv @ 000002a285413200] Non-monotonous DTS in output stream 0:0; previous: 283, current: 133; changing to 283. This may result in incorrect timestamps in the output file.
[flv @ 000002a285413200] Non-monotonous DTS in output stream 0:0; previous: 283, current: 200; changing to 283. This may result in incorrect timestamps in the output file.
[flv @ 000002a285413200] Non-monotonous DTS in output stream 0:0; previous: 283, current: 233; changing to 283. This may result in incorrect timestamps in the output file.
frame=22107 fps= 20 q=-1.0 size=  713271kB time=00:18:25.30 bitrate=5286.4kbits/s speed=   1x

-----------------

This is what I receive this repeating in yellow when something happens to the stream connection:

    [flv @ 0000026847b60040] Non-monotonous DTS in output stream 0:1; previous: 47046874, current: 7601881; changing to 47046874. This may result in incorrect timestamps in the output file.

    


    


    



    Great site for info and I have learned quite a bit so far in my quest to learn programming. There are a bunch of us who live stream to youtube and use ffmpeg. All we want to do is with the least amount of flags in our command line and make it simple to send a live rtsp connection from our cameras to youtube. I know we have some command line interpretations that we have wrong or need to be taken out or added. Could any of you assist with what we would need to resolve our yellow warnings and when it appears the rtsp stream crashes. I apologize with the learning curve for entering code into the post.

    


  • Transcode from a live m3u8 using -ss

    20 août 2015, par pgm

    I’m trying to create a VOD hls clip from a live hls stream on adobe media server using ffmpeg and nodejs.

    An example of the command I’m using looks like this :

    ffmpeg -report -analyzeduration 999999999 -probesize 999999999 -ss 50 -i http://live.m3u8 -y -r 29.97 -threads 0 -hls_list_size 0 -c:v copy -a:v copy streamoutput.m3u8

    The problem is the -ss param (start time) is calculating the start time from the live point on the stream, rather than from the first ’ts’ fragment. I’d like to be able to encode inside of a "DVR window," meaning seeking from the beginning of the stream, not from the live point of the stream.

    Example : I use the param -ss 50 and it won’t encode for 50 seconds until the live stream catches up, outputting this in the ffmpeg log :

    [h264 @ 0000000002beae00] non-existing PPS 0 referenced
    [h264 @ 0000000002beae00] non-existing PPS 0 referenced
    [h264 @ 0000000002beae00] decode_slice_header error

    Once the live stream catches up to the 50 second delay it begins encoding. It works this way when I use -ss as either an input parameter or output parameter.

    Is there a way to accomplish this ? I’ve noticed that if I leave -ss completely out of the command, it will start at the beginning of the stream, but as soon as it’s there, even as a 0, it will start at the "live point."

    Any help is much appreciated !

  • av1/h264_metadata : Don't reinitialize data

    20 juin 2019, par Andreas Rheinhardt
    av1/h264_metadata : Don't reinitialize data
    

    If the relevant elements (the color description elements for AV1 and the
    VUI elements in general for H.264 (since 1156b507)) are absent, then their
    correct values (usually meaning unknown) have already been inferred by
    the reading process, so that it is unnecessary to initialize them again
    in the av1/h264_metadata filters even when they were initially absent.

    Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>

    • [DH] libavcodec/av1_metadata_bsf.c
    • [DH] libavcodec/h264_metadata_bsf.c