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Autres articles (35)

  • 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 (...)

  • 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" (...)

  • 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

Sur d’autres sites (5620)

  • movenc : fix QT chapter track character encoding

    31 juillet 2014, par John Stebbins
    movenc : fix QT chapter track character encoding

    An encoding ("encd") box is required to tell QT that the string is UTF8
    • [DBH] libavformat/movenc.c

  • avcodec/aacenc : add strict bit rate control option

    25 avril 2023, par Jeremy Wu
    avcodec/aacenc : add strict bit rate control option

    In certain use cases, controlling the maximum frame size is critical. An
    
example is when transmitting AAC packets over Bluetooth A2DP.
    

    While the spec allows the packets to be fragmented (but UNRECOMMENDED),
    
in practice most headsets do not recognize nor reassemble such packets.
    

    In this patch, we allow setting `bit_rate_tolerance` to 0 to indicate
    
that the specified bit rate should be treated as an upper bound up to
    
frame level.
    

    Signed-off-by : Jeremy Wu <jrwu@chromium.org>
    • [DH] libavcodec/aacenc.c
    • [DH] libavcodec/options_table.h

  • How to completely remove track[1] with FFMpeg

    3 février 2017, par ether

    we have a raw stream that we transform into AAC-LC using FFMpeg.
    We need to completely remove the track[1] (metadata/ID3) from it.
    We tried following these ideas : How to remove ID3 audio tag image (or metadata) from mp3 with ffmpeg
    but it only strips the information from track[1] at best.

    Here are the tests done so far :

    [FFMPEG - Commands tested]

    ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -vn -codec:a libfdk_aac -b:a 49152 -f flv rtmp://REMOTE_URL -rtmp_live live   
    

    
ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -vn -codec:a libfdk_aac -b:a 49152 -map_metadata -1 -f flv rtmp://REMOTE_URL -rtmp_live live   
    

    
ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -map 0:a -codec:a libfdk_aac -b:a 49152 -map_metadata -1 -f flv rtmp://REMOTE_URL -rtmp_live live   
    

    
ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -vn -codec:a libfdk_aac -b:a 49152 -map_metadata:s:a -1 -f flv rtmp://REMOTE_URL -rtmp_live live  
    

    
ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -vn -codec:a libfdk_aac -b:a 49152 -map_metadata -1,g:-1,g -f flv rtmp://REMOTE_URL -rtmp_live live  
    

    
ffmpeg -hide_banner -f dshow -audio_buffer_size 250 -i "audio=Microphone" -vn -dn -codec:a libfdk_aac -b:a 49152 -f flv rtmp://REMOTE_URL -rtmp_live live

    [FFMPEG - Output]

    Guessed Channel Layout for  Input Stream #0.0 : stereo  
    
Input #0, dshow, from 'audio=Microphone':  
    
  Duration: N/A, start: 952026.572000, bitrate: 1411 kb/s  
    
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s  
    
[rtmp @ 000000000045fea0] Ignoring unsupported var reason  
    
Output #0, flv, to 'rtmp://REMOTE_URL':  
    
  Metadata:  
    
    encoder         : Lavf57.14.100  
    
    Stream #0:0: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 49 kb/s  
    
    Metadata:  
    
      encoder         : Lavc57.15.100 libfdk_aac  
    
Stream mapping:  
    
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (libfdk_aac))  
    
Press [q] to stop, [?] for help  
    
size=      26kB time=00:00:03.70 bitrate=  57.1kbits/s  

    [FFPlay]

    Input #0, hls,applehttp, from 'http://REMOTE_URL':  
    
  Duration: N/A, start: 79550.112778, bitrate: 50 kb/s  
    
  Program 0  
    
    Metadata:  
    
      variant_bitrate : 0  
    
    Stream #0:0: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 50 kb/s  
    
    Stream #0:1: Data: timed_id3 (ID3  / 0x20334449)

    We just need to completely remove track[1] altogether.
    Anyone has an idea how to accomplish this other than using -map_metadata -1, -vn, -map ?

    Thank you

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