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  • Gestion des droits de création et d’édition des objets

    8 février 2011, par

    Par défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

Sur d’autres sites (9934)

  • 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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