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  • MediaSPIP 0.1 Beta version

    25 April 2011, by

    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 (...)

  • HTML5 audio and video support

    13 April 2011, by

    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 (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 March 2010, by

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3); le plugin champs extras v2 nécessité par (...)

On other websites (10958)

  • Using FFMPEG to split a 16 channel audio input source into 4 seperate 4 channel audio feeds for streaming

    30 December 2019, by Mathew Knight

    I hope someone can help

    I am currently trying to split a 16ch Dante audio feed from a separate machine into 4 different audio streams that I can use to then TX via RTMP to Wowza for MPEG-DASH encoding, at present i am just trying to split them into files, I will add the RTMP streaming later.

    The biggest issue I am encountering at current is that FFMPEG is returning me this error from my input string

    Filter channelsplit:WR has an unconnected output

    here is my current input string

    ffmpeg -f dshow -i audio="Dante Via Receive (Dante Via)" -filter_complex "[0:a]channelsplit=channel_layout=hexadecagonal[FL][FR][FC][BL][BR][BC][SL][SR][TFL][TFC][TFR][TBL][TBC][TBR][WL][WR]" -map "[FL][FR][FC][BL]" 1-4.wav -map "[BR][BC][SL][SR]" 5-8.wav -map "[TFL][TFC][TFR][TBL]" 9-12.wav -map "[TBC][TBR][WL][WR]" 13-16.wav

    and here is the full FFMPEG output

    ffmpeg version git-2019-12-26-b0d0d7e Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 9.2.1 (GCC) 20191125
    
  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-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
    
  libavutil      56. 37.100 / 56. 37.100
    
  libavcodec     58. 65.100 / 58. 65.100
    
  libavformat    58. 35.101 / 58. 35.101
    
  libavdevice    58.  9.101 / 58.  9.101
    
  libavfilter     7. 69.101 /  7. 69.101
    
  libswscale      5.  6.100 /  5.  6.100
    
  libswresample   3.  6.100 /  3.  6.100
    
  libpostproc    55.  6.100 / 55.  6.100
    
Guessed Channel Layout for Input Stream #0.0 : stereo
    
Input #0, dshow, from 'audio=Dante Via Receive (Dante Via)':
    
  Duration: N/A, start: 103082.790000, bitrate: 1411 kb/s
    
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    
File '1-4.wav' already exists. Overwrite? [y/N] y
    
File '5-8.wav' already exists. Overwrite? [y/N] y
    
File '9-12.wav' already exists. Overwrite? [y/N] y
    
File '13-16.wav' already exists. Overwrite? [y/N] y
    
Filter channelsplit:WR has an unconnected output

    I’m also getting the issue where FFMPEG is guessing that the channel count is stereo, which is incorrect but i’m having problems figuring out how to define the input stream as 16ch’s of audio

    Any help with this would be greatly recieved

    Cheers

    M

  • FFMPEG detect silence command runs correctly but doses not give the silence duration

    7 January 2020, by Aizayousaf

    I have a .wav audio file and I need to extract silence/pause duration in this file. I’m using ffmpeg with silence detect filter but I’m unable to understand why its not giving silence duration with this file while it gives result with other files. Can anyone help me to understand the out given below that why its not showing detected silences.

    Input Command:

    ffmpeg -i "input.wav" -af silencedetect=noise=-30dB:d=0.5 -f null -

    OutPut

    ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
    
built with gcc 9.1.1 (GCC) 20190807
    
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. 31.100 / 56. 31.100
    
libavcodec     58. 54.100 / 58. 54.100
    
libavformat    58. 29.100 / 58. 29.100
    
libavdevice    58.  8.100 / 58.  8.100
    
libavfilter     7. 57.100 /  7. 57.100
    
libswscale      5.  5.100 /  5.  5.100
    
libswresample   3.  5.100 /  3.  5.100
    
libpostproc    55.  5.100 / 55.  5.100
    

    
Guessed Channel Layout for Input Stream #0.0 : stereo
    
Input #0, wav, from 'D:\Research\PhD\Carolina\AD\wav\media.io_Wakeman_Rhyne_001_01.wav':
    
Duration: 00:17:38.04, bitrate: 1411 kb/s
    
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Stream mapping:
    
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
    
Press [q] to stop, [?] for help
    
Output #0, null, to 'pipe:':

  • PyQt-thread. Get dynamicly output

    1 January 2020, by ZPro

    I use PyQt-thread for parallel conversion of mp3 files to aac via ffmpeg.
    Here is my code:

    class SubprocessThread(QThread):
    
    signal = pyqtSignal('PyQt_PyObject')
    

    
    def __init__(self, command, args):
    
        QThread.__init__(self)
    
        self.command = command
    
        self.args = args
    

    
    def __del__(self):
    
        self.wait()
    

    
    def run(self):
    
        output = subprocess.check_output('{0} {1}'.format(self.command, self.args), shell=True).split()
    
        self.signal.emit(output)

    And here is example of usage:

    threads = []
    

    
            for part in parts.keys():
    
                args = "-i \'{0}.mp3\' -c:a aac -b:a {1}k \'{2}.m4a\'".format(
    
                    os.path.join(tmp_dir, str(part)),
    
                    int(self.bitrate_cbx.currentText()),
    
                    os.path.join(tmp_dir, str(part)))
    
                print(args)  # debug
    
                ffmpeg_thread = SubprocessThread('ffmpeg', args)
    
                ffmpeg_thread.signal.connect(self.on_data_ready)
    
                threads.append(ffmpeg_thread)
    
                ffmpeg_thread.start()
    
                self.threads_count += 1

    I want to make progress bar, based on conversion, but ffmpeg always updates last string in his output (when conversion in progress).
    Here is an example of ffmpeg output while files are converting:

    user@host$ ffmpeg -i '/home/user/001.mp3' -c:a aac -b:a 128k -vn '/home/user/test.m4a' 
    
ffmpeg version n4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 9.2.0 (GCC)
    
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
    
  libavutil      56. 31.100 / 56. 31.100
    
  libavcodec     58. 54.100 / 58. 54.100
    
  libavformat    58. 29.100 / 58. 29.100
    
  libavdevice    58.  8.100 / 58.  8.100
    
  libavfilter     7. 57.100 /  7. 57.100
    
  libswscale      5.  5.100 /  5.  5.100
    
  libswresample   3.  5.100 /  3.  5.100
    
  libpostproc    55.  5.100 / 55.  5.100
    
Input #0, mp3, from '/home/user/001.mp3':
    
  Metadata:
    
    encoder         : Lavf57.41.100
    
    title           : test
    
    artist          : test
    
    album_artist    : test
    
    album           : test
    
    composer        : test
    
    genre           : test
    
    date            : 2018
    
  Duration: 00:12:38.02, start: 0.025056, bitrate: 192 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
    
    Metadata:
    
      encoder         : Lavc57.48
    
    Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 500x500 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    
    Metadata:
    
      comment         : Cover (front)
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> aac (native))
    
Press [q] to stop, [?] for help
    
Output #0, ipod, to '/home/user/test.m4a':
    
  Metadata:
    
    date            : test
    
    title           : test
    
    artist          : test
    
    album_artist    : test
    
    album           : test
    
    composer        : test
    
    genre           : test
    
    encoder         : Lavf58.29.100
    
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
    
    Metadata:
    
      encoder         : Lavc58.54.100 aac
    
size=   12107kB time=00:12:38.01 bitrate= 130.8kbits/s speed=79.2x

    How can I receive this data (string, that begins from "size=...") from my parallel QThreads to calculate overall progress?

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