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  • Librairies et logiciels spécifiques aux médias

    10 décembre 2010, par

    Pour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
    Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...)

  • 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 (...)

  • 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 (...)

Sur d’autres sites (9470)

  • How to set time_base when muxing AVI with libavformat ?

    18 mai 2012, par Anastasia

    In my application I receive from a remote server synchronized video
    (mpeg4) and audio (mp3) and mux them to avi file. The video comes
    frame-by-frame, and the audio comes in small chunks, say 200-250 ms.
    Both video frames and audio chunks have timestamps in milliseconds.

    My question is how to set audio time_base and pts correctly ?
    For video I set time_base.num = framerate, time_base.den = 1 ; and
    calculate pts as follows :

     AVRational time_base_1kHz;
     time_base_1kHz.num = 1;
     time_base_1kHz.den = 1000;
     packet.pts = av_rescale_q(timeStamp - baseTimeStamp_, time_base_1kHz, videoStream_->time_base);

    where baseTimeStamp_ is the 1st timestamp of the stream.

    But if I try do similar calculation for audio, I don't get playable
    avi. If I always set to audio AV_NOPTS_VALUE, then avi has playable
    video, but no audio.

    So what are the correct values for audio time_base and pts's ?
    Do I set time_base and pts's for video correctly ?

  • SOLVED - Compiling FFMPEG on Windows with Cywin and NDK r5

    19 mai 2012, par protectedmember

    This isn't a question - it's an answer for alll of you who have been facing the same problems as I have. I've been trying to compile this thing for a while now and I know of the numerous posts floating around the internet offering help. I have read and tried most of the suggestions and wanted to colate my success into this single post for others to benefit from.

    Since I don't have a blog, I thought it wouldn't hurt to post on here instead.

    I have managed to compile FFMPEG 0.10.3 (Freedom) on Windows 7 (32 bit) using NDK r5 and Cygwin. The steps :

    1 - Download/install Cygwin in the root of your C drive. I'm not going to give instructions on this, it's simple enough and there are plenty of tutorials on this.

    2 - Download NDK r5 from here and extract to the root of your C drive.

    3 - Download FFMPEG 0.10.3 from here and extract to the root of your C drive.

    4 - Open the file 'configure' in the root of the FFMPEG directory in a text editor.

    5 - Comment out lines 2073, 2074 and 2075.

    6 - Below 2075, add the following line :

    TMPDIR=c :/cygwin/tmp

    7 - Download this script (thankyou roman10) and place it inside your FFMPEG root directory. Rename the file to

    build_android.sh

    8 - Open the script in a text editor and edit line 17 to read

    c :/android-ndk-r5

    9 - Click start > run and type "bash" (without the speech marks) and press enter.

    10 - Type the following and press enter :

    cd /cygdrive/c/ffmpeg-0.10.3

    11 - Type the following and press enter :

    ./build_android.sh

    12 - Sit back and wait... libffmpeg.so will soon appear in your "c :\ffmpeg-0.10.3\android\" directory (where is defined in the bottom of the script from roman10's blog). The default architecture is armv7-a.

    The script from roman10's blog will actually compile quite a large shared object (.so) file. The compiler flags can be adjusted to suit your needs in the script from roman10's blog.

    I hope this helps,

    P.

  • How to send MPEGTS streams over UDP

    24 décembre 2015, par pAkY88

    I am developing a realtime video-streaming system which is composed basically by a server and several clients.

    For now, let’s ignore how packets are forwarded among the server and the clients, let’s focus just on how the server can send a MPEGTS stream over UDP packets.

    The stream is encoded in MPEGTS format.

    What I’m trying to do is reading some packets (the main question is "how many ?") and encapsulating them in UDP packets. The destination (a client) reads these UDP packets and then forward them to VLC, which is able to play MPEGTS network streams by reading UDP packets.

    If I send only video packets, everything works fine, instead if I try to encapsulate in the same UDP packet, both some video packets and some audio packets, VLC is not able to decode and play the stream.
    I read somewhere that each UDP packet should contain 7 TS packets, but unfortunately even if I comply with this rule, VLC doesn’t decode the stream correctly.

    Here is a sample code of my program : http://pastebin.com/evMi6FkY

    How should I encapsulate MPEGTS packets in UDP packets ?

    Thanks !