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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

Sur d’autres sites (12064)

  • Why the result of FFmpeg capture process has no audio to create a webm file ?

    23 mars 2019, par RAM

    The result of my bellow FFmpeg command has no audio (is silent) :

    ffmpeg -f gdigrab -framerate 30 -i desktop -video_size 720x480 -c:v libvpx-vp9 -c:a libopus -b:v 1M -b:a 128K -auto-alt-ref 0 -crf 10 -preset ultrafast output.webm

    But this one has audio :

    ffmpeg -f gdigrab -i desktop -f dshow -i audio="Microphone (4- High Definition Audio Device)" output.mkv
    • How should I capture as webm file by using libopus or libvorbis ?
    • What is the problem in my first command ?

    My FFmpeg version :

    ffmpeg version N-93439-gb073fb9eea Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8.2.1 (GCC) 20190212
    
  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-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-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
    
  libavutil      56. 26.100 / 56. 26.100
    
  libavcodec     58. 47.105 / 58. 47.105
    
  libavformat    58. 26.101 / 58. 26.101
    
  libavdevice    58.  7.100 / 58.  7.100
    
  libavfilter     7. 48.100 /  7. 48.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100

  • ffmpeg clean all noise background silences in a poscast

    23 mars 2019, par fireDevelop.com

    I have hundreds of podcast without music, just the voice and the room silence.
    In the silences, I have many clicks, respirations, etc...
    I need to clean all silences with a script, keeping intact the voice.

    In this picture you can see my dirty silences

    And here the result I want in all my audios

    When I use some scripts of sox. I don`t get the result I spect because the voice is affected by the script, the room-silence disappear and some clic still in the silences.

    Then in order to keep intact the voice, I want to do this :

    1. Delete all the silences longer than 3 seconds.

    1. Split all the audio and silences with in a sequence numbers. ie. :

      • 001-Silence-2.0seconds.wav
      • 002-voice.wav
      • 003-Silence-0.25seconds.wav
      • 004-voice.wav
      • 005-Silence-0.75seconds.wav
      • 006-voice.wav
      • ...
      • ...

    1. Before, run the script I created manually many files with silences of diferents silences I will use :

      • myManuallySilence-0.25seconds.wav
      • myManuallySilence-0.50seconds.wav
      • myManuallySilence-0.75seconds.wav
      • myManuallySilence-0.1seconds.wav
      • myManuallySilence-1.25seconds.wav
      • ...
      • ...
      • myManuallySilence-2.50seconds.wav
      • myManuallySilence-2.75seconds.wav
      • myManuallySilence-3.0seconds.wav
    1. the script will check the dirty silences duration and replace by the files myManuallySilence-x.xseconds.wav
    1. merge all files in one wav file, with the original voice and all the silences cleanned.

    At the moment I have only this script :

    # get the path of Adobe Audition and add timestamp in the output 
    
filename
    
fileName=out
    
current_time=$(date "+%Y.%m.%d-%H.%M.%S")
    
newFileName=$fileName.$current_time.wav
    
#yourPathAPP=/Applications/Adobe\ Audition\ CC\ 2019/Adobe\ Audition\ 
    
CC\ 2019.app
    
yourPathAPP=/Volumes/6TB/Applications/ocenaudio.app
    
# # First denoise audio
    

    
# ## Get noise sample
    
 ffmpeg -i in.wav -vn -ss 00:00:00 -t 00:00:01 noise-sample.wav
    

    
# ## Create noise profile
    
 sox noise-sample.wav -n noiseprof noise.prof
    

    
# ## Clean audio from noise
    
 sox in.wav $newFileName noisered noise.prof 0.50
    
# # Split audio by noise
    
 sox -V3 $newFileName output.wav silence 1 00:00:02.000 - 80d 1 
    
 00:00:02.000 -80d : newfile : restart
    

    
# ####### (these settings worked for my computer mic - maybe we need to 
    
finetune them later) #######

    Is getting all the voice in separate files like this :
    output001.wav
    output002.wav
    output003.wav
    output004.wav
    ...
    output00x.wav

    Please, any suggestion will be appreciated.
    Thanks so much in advance !

  • Transcoding to H264. PTS and DTS sync accross multiple output streams with different bitrates

    25 mars 2019, par timmytimmers

    I have a setup where I am transcoding live feeds from OTA broadcasts to H264 using the Nvidia NVENC encoder. I am also transcoding the audio to AAC. We are trying to output 3 cbr streams and various bitrates. The problem I am running into is that the PTS and DTS on the multiple outputs are not aligning which is critical for our use case. I am hoping there is an easy fix to this but I have not yet been able to locate one. Any thoughts on how to accomplish this ?

    ===> Source Feed <===

    ffprobe udp://@238.224.1.5:59005
    
ffprobe version N-93005-gd92f06e Copyright (c) 2007-2019 the FFmpeg developers
    
  built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
    
  configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/circle/ffmpeg_build/include --extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/circle/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc
    
  libavutil      56. 26.100 / 56. 26.100
    
  libavcodec     58. 44.100 / 58. 44.100
    
  libavformat    58. 26.100 / 58. 26.100
    
  libavdevice    58.  6.101 / 58.  6.101
    
  libavfilter     7. 48.100 /  7. 48.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
[mpeg2video @ 0x558e5a80fa40] Invalid frame dimensions 0x0.
    
    Last message repeated 22 times
    
Input #0, mpegts, from 'udp://@238.224.1.5:59005:
    
  Duration: N/A, start: 89037.540778, bitrate: N/A
    
  Program 3 
    
    Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
    
    Stream #0:1[0x34](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, 5.1(side), fltp, 384 kb/s
    
    Stream #0:2[0x35](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 192 kb/s

    ===> Command I am currently running to transcode <===

    screen -d -m ffmpeg -i 'udp://@238.224.1.5:59005?fifo_size=1000000&amp;overrun_nonfatal=1' \
    
        -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 6000K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test6000" -metadata service_provider="test" 'udp://@239.1.1.1:59001?pkt_size=1316' \
    
        -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 3500K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test3500" -metadata service_provider="test" 'udp://@239.1.1.2:59002?pkt_size=1316' \
    
        -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 1500K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test1500" -metadata service_provider="test" 'udp://@239.1.1.3:59003?pkt_size=1316'

    These streams will be eventually mux’d back together for DRM insertion into a ABR stream. Without those values being in sync it will not be ABR compliant.

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