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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (11799)

  • FFMPEG : How to merge two sounds with good quality ?

    26 avril 2021, par Mahmoud Eidarous

    I'm trying to merge a recorded sound with an audio file and make it sound like an original song.

    


    I'm using this command

    


    ffmpeg -i audiofile.mp3 -i recorded_file.wav -filter_complex "[0:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=0.5[a1]; [1:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=7[a2]; [a1][a2]amerge=inputs=2,pan=stereo|c0code>

    


    This command does achieve the merge, but the quality of the output song is very bad.

    


    I tried to manipulate the volume values but still the quality is bad.

    


    I believe a filter is required on the recorded sound or something...

    


    Any help to make the output sound like an original song would be really appreciated.

    


  • qsv : Make the hevc idr_interval consistent with the h264 one

    17 octobre 2017, par Luca Barbato
    qsv : Make the hevc idr_interval consistent with the h264 one

    According to the MediaSDK documentation the idr_interval value has
    
a different meaning depending on the codec :
    

        0 in H264 means make every I-frame IDR, in HEVC it means to have
    
    it only at the beginning.
    

        1 in H264 means every other I-frame is not-IDR, in HEVC it means
    
    that every I-frame is IDR.
    

    Keep the behaviour consistent between the two encoders by increasing
    
by 1 internally the idr_interval value for HEVC.
    • [DBH] libavcodec/qsvenc_hevc.c

  • Why does libmp3lame add zeros to the start of the MP3 ?

    29 mars 2016, par Phlox Midas

    I have a uncompressed .wav file that I turn into a 96k MP3 file :

    ffmpeg.exe -i song.wav -vn -b:a 96000 -ac 2 -ar 48000 -acodec libmp3lame -y song.mp3

    The input file has 637386 samples. The output has 639360 samples. The extra samples in the MP3 are all zeros at the beginning of the file. This happens in every file I’ve translated and with more codecs than just libmp3lame. Is this an FFMPEG bug or a codec bug ? Why are these added ? Is there a way to stop them from being added ?

    Edit : Simplified example and console output :

    ffmpeg.exe -i song.wav -y song.mp3
    

    
ffmpeg version N-55796-gb74213d Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Aug 26 2013 19:43:51 with gcc 4.7.3 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
    
  libavutil      52. 42.100 / 52. 42.100
    
  libavcodec     55. 29.100 / 55. 29.100
    
  libavformat    55. 14.102 / 55. 14.102
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 82.102 /  3. 82.102
    
  libswscale      2.  5.100 /  2.  5.100
    
  libswresample   0. 17.103 /  0. 17.103
    
  libpostproc    52.  3.100 / 52.  3.100
    
Guessed Channel Layout for  Input Stream #0.0 : stereo
    
Input #0, wav, from 'song.wav':
    
  Duration: 00:00:13.28, bitrate: 1538 kb/s
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
    
Output #0, mp3, to 'song.mp3':
    
  Metadata:
    
    TSSE            : Lavf55.14.102
    
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (pcm_s16le -> libmp3lame)
    
Press [q] to stop, [?] for help
    
size=     208kB time=00:00:13.29 bitrate= 128.4kbits/s
    
video:0kB audio:208kB subtitle:0 global headers:0kB muxing overhead 0.111205%

    Number of samples in wav : 637386

    Number of samples in mp3 : 639984

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