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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 (...)

  • 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 (...)

  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, 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 (...)

Sur d’autres sites (11840)

  • Can I know which byte range to read from a remote mp4 file for FFMpeg to decode a keyframe ?

    12 octobre 2023, par db9117

    I need to decode a of keyframe of a video file (mp4, h264 encoded). I know the timestamp of the keyframe I want to extract/decode. I want to minimize amount of data being read in memory. For this, I need to know beforehand exactly the minimal byte range I would require that encompasses this keyframe. How do I know what is the minimal byte range in the whole mp4 byte stream I need to read in order to be able to decode the keyframe ?

    


    I currently find the appropriate keyframe in the index_entries contained in the header. I get its byte position (pos attribute) and timestamp (timestamp attribute). I calculate the range as follows :

    


    startBytes : minimum of :

    


      

    1. the pos of the keyframe
    2. 


    3. the pos of the nearest index entry in the audio stream happening at or before the keyframe's timestamp.
    4. 


    


    This way when it's decoding the frame, if it also needs the audio content for demuxing, it would have it.

    


    endBytes : maximum of :

    


      

    1. the pos of the next frame in the video stream's index, after the keyframe
    2. 


    3. the pos of the next frame in the audio stream's index after the timestamp of the wished keyframe.
    4. 


    


    This way I know that I have everything up until the next frame in the index, which theoretically should be enough to decode the keyframe only.

    


    I then read the appropriate byte range.

    


    When I try to decode the frame, I run in a loop until I succeed :

    


      

    • avcodec_read_frame
    • 


    • avcodec_send_packet
    • 


    • avcodec_receive_frame
    • 


    


    I ignore AVERROR(EAGAIN) errors.

    


    avcodec_receive_frame fails multiple times with error AVERROR(EAGAIN) which I ignore, until it fails saying that the memory it wants to read isn't available (wants to read after endBytes). I explicitly tell it to fail if it wants to read more than it has already read.

    


    Note : for other keyframes at other positions in other videos, it sometimes succeeds (probably because the range is big enough by chance), but it fails more often than not.

    


    My question is : Why is the end of the range not enough to be able to decode only the one keyframe ? Is there any way to more precisely calculate the exact range in bytes I would need in order to decode a particular keyframe ?

    


  • mp3 to wav with ffmpeg reduces quality and duration

    25 novembre 2020, par Neret

    I took this mp3 file and converted it to wav with Audition and with this ffmpeg command :

    


    ffmpeg -i "Casey Don’t You Fret - Dan Lebowitz.mp3" -c:a pcm_f32le "Casey Don’t You Fret - Dan Lebowitz_FFMPEG.wav"


    


    After that I checked the statistics in Audition. The wav file which was generated with Audition has exactly the same statistics as the original mp3 file.

    


    enter image description here

    


    The duration of ffmpeg file has changed. Audio statistics became worse.
Why is this happening ? Can I fix it ?

    


    I used ffmpeg version 2020-11-22-git-0066bf4d1a-full_build-www.gyan.dev on Windows.

    


    UPDATE 1 :
I cut a few seconds of mp3 at the beginning and at the end :

    


    enter image description here

    


    FFMPEG added silence at the beginning and increased duration.

    


    UPDATE 2 :
Look how ffmpeg changed the waveform of this 2.mp3 file in the middle :

    


    ffmpeg -y -i 2.mp3 -c:a pcm_f32le 2_FFMPEG.wav


    


    enter image description here
enter image description here

    


  • TCP Connection refused error when using FFMPEG for audio stream to HTTP on macOS

    26 novembre 2020, par freddy

    I'm trying to stream my microphone input via HTTP using ffmpeg, so I can stream it in HTML. I run the following ffmpeg command :

    


    ffmpeg -f avfoundation -i ":1" -c:a libmp3lame -f mp3 -r 30 http://localhost:809

    


    It, however, crashes with the following error message :

    


    ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.27)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Input #0, avfoundation, from ':1':
  Duration: N/A, start: 3445.340045, bitrate: 22579 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, hexadecagonal, flt, 22579 kb/s
[tcp @ 0x7fa46ec96600] Connection to tcp://localhost:8090 failed: Connection refused
http://localhost:8090: Connection refused


    


    I've had success with streaming on that port using VLC, but it for some reason won't work using ffmpeg. Any ideas on how to fix this ?