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  • Le plugin : Podcasts.

    14 juillet 2010, par

    Le problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
    Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
    Types de fichiers supportés dans les flux
    Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

  • Taille des images et des logos définissables

    9 février 2011, par

    Dans beaucoup d’endroits du site, logos et images sont redimensionnées pour correspondre aux emplacements définis par les thèmes. L’ensemble des ces tailles pouvant changer d’un thème à un autre peuvent être définies directement dans le thème et éviter ainsi à l’utilisateur de devoir les configurer manuellement après avoir changé l’apparence de son site.
    Ces tailles d’images sont également disponibles dans la configuration spécifique de MediaSPIP Core. La taille maximale du logo du site en pixels, on permet (...)

Sur d’autres sites (6297)

  • Can one "fix" a video with non monotonic DTS ?

    30 novembre 2020, par amn

    I've got a number of WebM containers (input.webm), each with a VP9 video track and an Opus audio track. These containers (files) were generated with an older version of Google Chrome using the MediaRecorder Web API, by a live media recording Web application.

    


    As part of an attempt to make the material available to a certain overly stubborn Windows DirectShow application that is otherwise refusing to import WebM files — despite the fact that we've got DirectShow filters for demuxing WebM and decoding VP9 and Opous installed and verified to be working on the application host — I have tried to introduce a re-muxing batch job where the media is re-muxed into an MPEG-4 container, stuffing the same VP9 and Opus tracks in the MP4 file, for intended consumption by said Windows application.

    


    However, on at least one of these WebM files, FFmpeg refuses to proceed with generating the output, using ffmpeg -i input.webm -c copy output.mp4 :

    


    [mp4 @ 00000208bee17840] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 53744 >= 53744
av_interleaved_write_frame(): Invalid argument


    


    Given how some of the material in the WebM collection are quite lengthy (on the order of hours), I don't want to transcode video (or audio, for that matter) tracks, instead opting to try and stuff them into an MPEG-4 container. As far as I understand, VP9 in MPEG-4 is rather well supported, too — certainly FFmpeg has been able to re-mux most of the WebM files we've got into MPEG-4 containers which the aforementioned application has been able to import, too.

    


    Is there anything I can do to "fix" [WebM] files where DTS is out of order ? Is this some error during original encoding of the media file in question ? Google Chrome does use FFmpeg for encoding media, by the way, at least I know earlier versions did.

    


  • Ffmpeg [NULL @ 0x56390335ae80] Unable to find a suitable output format for 'listen' listen : Invalid argument

    18 février 2021, par Sowmya

    I am trying to record a live stream using ffmpeg. I am using an sdp file as input and trying to put it in a .mp3 file and i get the following error :

    


    ffmpeg listen  -loglevel debug -protocol_whitelist file,crypto,udp,rtp,tcp  -i audio.sdp -f mp3 output.mp3
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Splitting the commandline.
Reading option 'listen' ... matched as output url.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp,tcp'.
Reading option '-i' ... matched as input url with argument 'audio.sdp'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'output.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url audio.sdp.
Successfully parsed a group of options.
Opening an input file: audio.sdp.
[NULL @ 0x5639033039c0] Opening 'audio.sdp' for reading
[sdp @ 0x5639033039c0] Format sdp probed with size=2048 and score=50
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 48000
[sdp @ 0x5639033039c0] audio channels set to: 2
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 16000
[sdp @ 0x5639033039c0] audio channels set to: 1
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 32000
[sdp @ 0x5639033039c0] audio channels set to: 1
[sdp @ 0x5639033039c0] audio codec set to: opus
[sdp @ 0x5639033039c0] audio samplerate set to: 8000
[sdp @ 0x5639033039c0] audio channels set to: 1
[udp @ 0x56390330b3c0] end receive buffer size reported is 131072
[udp @ 0x56390330b5e0] end receive buffer size reported is 131072
[sdp @ 0x5639033039c0] setting jitter buffer size to 500
[sdp @ 0x5639033039c0] Before avformat_find_stream_info() pos: 1281 bytes read:1281 seeks:0 nb_streams:1
[sdp @ 0x5639033039c0] After avformat_find_stream_info() pos: 1281 bytes read:1281 seeks:0 frames:0
Input #0, sdp, from 'audio.sdp':
  Metadata:
    title           : -
  Duration: N/A, bitrate: N/A
    Stream #0:0, 0, 1/8000: Audio: opus, 48000 Hz, mono, fltp
Successfully opened the file.
Parsing a group of options: output url listen.
Successfully parsed a group of options.
Opening an output file: listen.
[NULL @ 0x56390335ae80] Unable to find a suitable output format for 'listen'
listen: Invalid argument
[AVIOContext @ 0x56390330c6c0] Statistics: 1281 bytes read, 0 seeks


    


    I have tried with output as .wav file too, but i get the same error, Please help me out. Thanks in advance !

    


  • Audio : Retrieve bitdepth with ffmpeg api

    22 août 2021, par User42

    I need the bitdepth of arbitrary audio formats
(mp3, wav, acc, flac, opus, ogg etc.)

    


    For that I tried

    


    AVStream.codecpar.bits_per_raw_sample ;

    


    and also

    


    AVStream.codec.bits_per_raw_sample ;

    


    But except of flac all return 0.

    


    ffprobe also doesn't output the bitdepth,
only the sample format (s16, fltp etc.).

    


    How do I get the bitdepth ?

    


    Or do I have to "derive" it from sample format ?

    


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