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  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • 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

  • 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 (...)

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  • Android receive RTP/UDP audio stream from VLC/ffmpeg

    23 mai 2016, par Dom4S

    I was searching for a good answer for half a day, but I am a beginner at this stuff and I would appreciate any help.

    What I would like to achieve is to stream audio (mp3 files) within ffmpeg or vlc and receive it on an Android device by udp/rtp.

    This is what I was able to figure out myself sofar :

    1) There is Android class AudioStream and RTPStream. What I don’t know is how to use it. For example I create a stream via ffmpeg with : ffmpeg -re -i mymp3.mp3 -ar 8000 -acodec copy -f rtp rtp://192.168.0.100:5533, where 192.168.0.100 is the address of my Android device. Now I would like to receive it and play it.

    I found something like this on Stack :

    AudioStream audioStream;
    AudioGroup audioGroup;
    AudioCodec codec = AudioCodec.PCMU;
    StrictMode.ThreadPolicy policy = new StrictMode.ThreadPolicy.Builder().permitNetwork().build();
    StrictMode.setThreadPolicy(policy);
    AudioManager audio = (AudioManager)getSystemService(AUDIO_SERVICE);
    audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
    audioGroup = new AudioGroup();
    audioGroup.setMode(AudioGroup.MODE_NORMAL);
    InetAddress inetAddress;
    try {
       inetAddress = InetAddress.getByName("163.11.62.208");
       audioStream = new AudioStream(inetAddress);
       audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
       audioStream.setCodec(codec);
       InetAddress inetAddressRemote = InetAddress.getByName("163.11.169.206");
       audioStream.associate(inetAddressRemote, 5004);
       audioStream.join(audioGroup);
    }

    What is the first inetAddress 163.11.62.208 and what is the second one 163.11.169.206 ? Shoudln’t I just give an address of a stream ?

    2) Can I submit only streams in PCMU format ? Can I stream mp3 files ?

    3) Is it even possible ?

  • FFMPEG : Using av_write_frame AFTER av_write_trailer

    20 mars 2024, par Aleksei Komarov

    I'm using avio_alloc_context with my own "write_buffer_proc" to override file i/o.
I am trying to continue writing the stream with av_write_frame AFTER av_write_trailer was called.
But av_write_trailer has some strange side effects.

    


    The result is AV in module 'avformat-58.dll' (Read of address 0000000000000090)
I think that FormatContext becomes invalid after av_write_trailer.
How Can I fix it ?
Can I try to save FormatContext somewhere before calling av_write_trailer ?

    


  • How do I allow MPEG tcp 'pull' from FFMPEG

    7 mars 2021, par Ian

    I am using FFMPEG to create a TCP MPEG-2 stream using the following inside of nginx :

    


    exec ffmpeg -i rtmp ://localhost:1935/live/$name -codec copy -g 1 -bsf:v h264_mp4toannexb -f mpegts tcp ://192.168.1.225:1235 ;

    


    I'd like to allow a client to pull a stream from this server rather than pushing to a specific address. Can I do this ?