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  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Problèmes fréquents

    10 mars 2010, par

    PHP et safe_mode activé
    Une des principales sources de problèmes relève de la configuration de PHP et notamment de l’activation du safe_mode
    La solution consiterait à soit désactiver le safe_mode soit placer le script dans un répertoire accessible par apache pour le site

  • Encodage et transformation en formats lisibles sur Internet

    10 avril 2011

    MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
    Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
    Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...)

Sur d’autres sites (8635)

  • ffmpeg set buf_len

    2 avril 2013, par Jeetendra_Nath_Jha

    I have been taken input from the device file /dev/video49.

    using ffmpeg, following command writing from the terminal :

    ffmpeg -f video4linux2 -i /dev/video49 -target ntsc-vcd -s 1920x1080 pixfmt:yuv420p -vstats -bufsize 225k -an -f mpegts -vcodec mpeg4 udp://127.0.0.1:10000?pkt_size=1316

    but it gives an error given below :

    ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
     built on Mar 20 2013 15:06:14 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
     configuration:
     libavutil      52. 18.100 / 52. 18.100
     libavcodec     54. 92.100 / 54. 92.100
     libavformat    54. 63.104 / 54. 63.104
     libavdevice    54.  3.103 / 54.  3.103
     libavfilter     3. 42.103 /  3. 42.103
     libswscale      2.  2.100 /  2.  2.100
     libswresample   0. 17.102 /  0. 17.102
    [video4linux2,v4l2 @ 0x95862a0] buf_len[0] = 1048576 < expected frame size 3110400
    /dev/video49: Cannot allocate memory

    I don't know how to set the buf_len here...

    because in an error it says that the buf_len is need to be set.

  • Error compiling and building DashEncoder code and how to play the .mpd file when generated [closed]

    11 avril 2013, par niuu

    I'm trying to build the DashEncoder code which I downloaded from github https://github.com/slederer/DASHEncoder. Well, I followed all the instructions given in the how to compile dash file. installed Gpac n X264 and compiled both successfully. Then did make of Dashencoder and ran it as ./Dashencoder. But I found some issues in it. Got this log :

    ==========DASH ENCODER===============
    Unknown option in resourcefile : sql-pw :
    current encoder x264
    YES
    x264 encoding @ 300 kbps : Pass 1
    x264 : x264 —profile baseline —preset slow —verbose —fps 24 —vbv-maxrate 300 —vbv-bufsize 600 —scenecut 0 —keyint 48 —output /opt/lampp/htdocs/tests_updates/sintel_trailer_2k_480p24_300kbit.h264 /home/niu/sintel_trailer_2k_480p24.y4m >out.txt 2>&1
    mkdir : cannot create directory

    /opt/lampp/htdocs/tests_updates/sintel_300kbit': File exists
    cp: omitting directory

    /opt/lampp/htdocs/tests_updates/'
    copy audio : cp /opt/lampp/htdocs/tests_updates/ /opt/lampp/htdocs/tests_updates/sintel_300kbit/MP4Box multiplexing Video : /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.h264
    mp4box : MP4Box -add /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.h264 /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4
    AVC-H264 import - frame size 854 x 480 at 24.000 FPS
    AVC Import results : 1253 samples - Slices : 27 I 1226 P 0 B - 1 SEI - 27 IDR
    Saving to /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4 : 0.500 secs Interleaving
    MP4Box multiplexing Audio :/opt/lampp/htdocs/tests_updates/sintel_300kbit/
    mp4box : MP4Box -add /opt/lampp/htdocs/tests_updates/sintel_300kbit/ /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4
    Unknown input file type
    Unknown input file type
    Error importing /opt/lampp/htdocs/tests_updates/sintel_300kbit/ : Bad Parameter
    MP4Box Cleaning ...
    mp4box : MP4Box -no-sys /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4
    Saving /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4 : 0.500 secs Interleaving
    MP4Box segmentation : /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.h264
    mp4box : MP4Box -frag 2000 -dash 2000 -rap -segment-name /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4
    DASH-ing file : 2.00s segments 2.00s fragments single sidx per segment
    Spliting segments at GOP boundaries
    [DASH] Generating MPD at time 2013-03-16T16:40:03Z
    DASHing file /opt/lampp/htdocs/tests_updates/sintel_300kbit/sintel_trailer_2k_480p24_300kbit.mp4
    terminate called after throwing an instance of 'std::out_of_range'
    what() : basic_string::substr
    Error : Unable to open MPD file !Aborted
    Why is this error at the end ? and also one folder got created in my /opt/lampp/htdocs/tests_updates/sintel_300kbit. which has two types of files : 27 files of .m4s extension 1 file -sintelinit.mp4 1 file- sintel_trailer_2k_480p24_300kbit.mp4 , which when played in vlc player played the video bt no audio ! &

    1 file- sintel_trailer_2k_480p24_300kbit.h264 which cannot be opened.

    No .mpd file was created.

    Also I want to know aft creating that .mpd file how will i be able to test it on my android client say media player.

    I am damn confused with all this happening. Please help

  • using pocketsphinx_continuous with a .wav file

    3 avril 2013, par user2242131

    I am attempting to write an application that will allow a user to speak a small set of commands from a remote system and have them executed on my server. Using pocketsphinx to parse the spoken text. When run locally with the microphone, pocketsphinx_continuous works perfectly no matter how I slur the words. But when importing the audio file and using ffmpeg to downsample the audio to a single channel, 16 bit PCM file, it will parse the first word without difficulty. Then it will skip everything else and treat it as . I am confident that the problem is in the file format and not in the pocketsphinx configuration.

    Using command line
    ffmpeg -y -i Sound\AddSheet.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav
    in a batch file.

    The bottom of the output I get is :

    INFO: fsg_search.c(1407): Start node ADD.0:5:47
    INFO: fsg_search.c(1407): Start node <sil>.0:2:49
    INFO: fsg_search.c(1446): End node <sil>.126:128:305 (-486)
    INFO: fsg_search.c(1662): lattice start node <s>.0 end node <sil>.126
    INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:126:305) = -175371
    INFO: ps_lattice.c(1390): Joint P(O,S) = -176076 P(S|O) = -705
    000000000: ADD USER
    </sil></sil></s></sil></sil>

    Which is not the audio in the file. The words spoken in the file are "ADD SPREADSHEET", which works perfectly from the same microphone without the intervening .wav file.

    I have tried increasing the audio volume and decreasing the background noise using sox :

    sox -v 3.0 Sound\%1 Sound\%1-loud.wav ffmpeg -i Sound\%1-loud.wav -vn -ss 00:00:00 -t 00:00:01 -y Sound\%1-noiseaud.wav
    sox Sound\%1-noiseaud.wav -n noiseprof Sound\%1-noise.prof
    sox Sound\%1 Sound\%1-clean.wav noisered sound\noise.prof 0.21
    ffmpeg -y -i Sound\%1-clean.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav

    with no noticeable effect on the final results.

    If you look at the output you will notice that fsg_search.c has found ADD as the start node, then silence for the remainder. Please help on this.