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  • Participer à sa traduction

    10 avril 2011

    Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
    Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
    Actuellement MediaSPIP n’est disponible qu’en français et (...)

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • 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 (...)

Sur d’autres sites (5093)

  • How to transmux ismv file to mp4

    21 septembre 2013, par Priyal

    I wanted to convert ismv file to mp4 without any packet loss.

    I tried it with ffmpeg as :

    ffmpeg -i input.ismv output.mp4

    But the resultant mp4 file has some packet/frame loss due which video is distracted.

    ffmpeg command line return the following :

    Metadata: major_brand : isml minor_version : 1 compatible_brands: piffiso2 Duration: 00:03:57.18, start: 0.000000, bitrate: 1612 kb/s
    Stream #0:0(und): Audio: wmapro (b[1][0][0] / 0x0162), 44100 Hz, stereo, fltp, 64 kb/s (default)Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Audio

    Stream #0:1(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 1343 kb/s, 30 tbr, 10000k tbn, 60 tbc (default) Metadata: creation_time : 2013-09-20 06:18:40

    Video Stream #0:2(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 480x360 [SAR 1:1 DAR 4:3], 887 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (default) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video

    Stream #0:3(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 364x27 2 [SAR 136:136 DAR 91:68], 599 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (default ) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video

    Stream #0:4(und): Video: vc1 (Advanced) (vc-1 / 0x312D6376), yuv420p, 276x20 8 [SAR 207:208 DAR 14283:10816], 400 kb/s, 30 fps, 30 tbr, 10000k tbn, 60 tbc (d efault) Metadata: creation_time : 2013-09-20 06:18:40 handler_name : Video

    [wmapro @ 003fa000] Channel transform bit is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.

    Error while decoding stream #0:0: Invalid data found when processing input [wmapro @ 003fa000] frame[0] would have to skip -9 bits[wmapro @ 003fa000] Packet loss detected! seq e vs d [wmapro @ 003fa000] Packet loss detected! seq d vs f [wmapro @ 003fa000] Reserved bit is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented
  • FFmpeg : no image is displayed when broadcasting flv made of speex / mellymoser and libx264

    15 septembre 2013, par wvxvw
    ffmpeg -f alsa -i hw:PCH,0 -f x11grab -s 1920x1200 -r 15 -i :0.0 \
       -acodec nellymoser -ar 22050 -ac 1 -audio_preload 1000000 -vcodec libx264 \
       -preset fast -pix_fmt yuv420p -s 1280x800 -maxrate 1000k -threads 0 -f flv "$URL"

    This is the command I'm using, but I don't get any visible output, only sound. FFmpeg gives no warnings. What am I doing wrong ?


    If I remove -audio_preload, the streaming works, but the audio comes with increasing delay (grows about a minute every minute).


    The output, as per request :

    ffmpeg version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
     built on May 14 2013 21:59:35 with gcc 4.7.2 (GCC) 20121109 (Red Hat 4.7.2-8)
     configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
     libavutil      51. 73.101 / 51. 73.101
     libavcodec     54. 59.100 / 54. 59.100
     libavformat    54. 29.104 / 54. 29.104
     libavdevice    54.  2.101 / 54.  2.101
     libavfilter     3. 17.100 /  3. 17.100
     libswscale      2.  1.101 /  2.  1.101
     libswresample   0. 15.100 /  0. 15.100
     libpostproc    52.  0.100 / 52.  0.100
    [alsa @ 0xc5e8e0] Estimating duration from bitrate, this may be inaccurate
    Guessed Channel Layout for  Input Stream #0.0 : stereo
    Input #0, alsa, from 'hw:PCH,0':
     Duration: N/A, start: 1379234523.677560, bitrate: 1536 kb/s
       Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    [x11grab @ 0xc6cda0] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1920 height: 1200
    [x11grab @ 0xc6cda0] shared memory extension found
    [x11grab @ 0xc6cda0] Estimating duration from bitrate, this may be inaccurate
    Input #1, x11grab, from ':0.0':
     Duration: N/A, start: 1379234523.843273, bitrate: 1105920 kb/s
       Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1920x1200, 1105920 kb/s, 15 tbr, 1000k tbn, 15 tbc
    [libx264 @ 0xc59e00] VBV maxrate specified, but no bufsize, ignored
    [libx264 @ 0xc59e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
    [libx264 @ 0xc59e00] profile High, level 3.2
    [libx264 @ 0xc59e00] 264 - core 128 r2223 f6a8615 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=15 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    Output #0, flv, to 'rtmp://live.justin.tv/app/live_key':
     Metadata:
       encoder         : Lavf54.29.104
       Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 1280x800, q=-1--1, 1k tbn, 15 tbc
       Stream #0:1: Audio: nellymoser ([6][0][0][0] / 0x0006), 22050 Hz, mono, flt, 128 kb/s
    Stream mapping:
     Stream #1:0 -> #0:0 (rawvideo -> libx264)
     Stream #0:0 -> #0:1 (pcm_s16le -> nellymoser)
    Press [q] to stop, [?] for help
    [alsa @ 0xc5e8e0] ALSA buffer xrun.
  • Desktop audio falls behind when recording microphone + desktop audio + screen using ffmpeg

    15 septembre 2013, par madr

    I have put together this script for recording the microphone, the desktop audio and the screen using ffmpeg :

    DATE=`which date`
    RESO=2560x1440
    FPS=30
    PRESET=ultrafast
    DIRECTORY=$HOME/Video/
    FILENAME=videocast`$DATE +%d%m%Y_%H.%M.%S`.mkv

    ffmpeg -y -vsync 1 \
    -f pulse -ac 2 -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor \
    -f pulse -ac 1 -ar 25000 -i alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono \
    -filter_complex aresample=async=1,amix=duration=shortest,apad \
    -f x11grab -r $FPS -s $RESO -i :0.0 \
    -acodec libvorbis \
    -vcodec libx264 -pix_fmt yuv420p -preset $PRESET -threads 0 \
    $DIRECTORY$FILENAME

    Everything is recorded and between the screen and the microphone sound there are no issues what so ever, however the desktop audio falls behind badly.

    It begins in sync but gets worse over time during playback, also in ffplay. It does not matter what application playing sound : both Youtube-videos in the browser, desktop sounds and Rhythmbox (playing a couple of seconds of song then stops, wait and repeat) gets out of sync.

    The terminal output complain about

    "ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred"

    and similar but I do not know what that means.

    Full terminal output here :

    ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
     built on Aug 11 2013 14:52:28 with gcc 4.8.1 (GCC) 20130725 (prerelease)
     configuration: --prefix=/usr --disable-debug --disable-static --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic --enable-postproc --enable-runtime-cpudetect --enable-shared --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab
     libavutil      52. 38.100 / 52. 38.100
     libavcodec     55. 18.102 / 55. 18.102
     libavformat    55. 12.100 / 55. 12.100
     libavdevice    55.  3.100 / 55.  3.100
     libavfilter     3. 79.101 /  3. 79.101
     libavresample   1.  1.  0 /  1.  1.  0
     libswscale      2.  3.100 /  2.  3.100
     libswresample   0. 17.102 /  0. 17.102
     libpostproc    52.  3.100 / 52.  3.100
    Guessed Channel Layout for  Input Stream #0.0 : stereo
    Input #0, pulse, from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor':
     Duration: N/A, start: 0.014093, bitrate: 1536 kb/s
       Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    Guessed Channel Layout for  Input Stream #1.0 : mono
    Input #1, pulse, from 'alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono':
     Duration: N/A, start: 0.006172, bitrate: 400 kb/s
       Stream #1:0: Audio: pcm_s16le, 25000 Hz, mono, s16, 400 kb/s
    [x11grab @ 0x218a6e0] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 2560 height: 1440
    [x11grab @ 0x218a6e0] shared memory extension found
    Input #2, x11grab, from ':0.0':
     Duration: N/A, start: 1379021580.184321, bitrate: N/A
       Stream #2:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 2560x1440, -2147483 kb/s, 30 tbr, 1000k tbn, 30 tbc
    [libx264 @ 0x21ae560] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    [libx264 @ 0x21ae560] profile Constrained Baseline, level 5.0
    [libx264 @ 0x21ae560] 264 - core 133 r2339 585324f - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
    Output #0, matroska, to '/home/anders/Video/videocast12092013_23.33.00.mkv':
     Metadata:
       encoder         : Lavf55.12.100
       Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 25000 Hz, mono, fltp
       Stream #0:1: Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 2560x1440, q=-1--1, 1k tbn, 30 tbc
    Stream mapping:
     Stream #0:0 (pcm_s16le) -> aresample (graph 0)
     Stream #1:0 (pcm_s16le) -> amix:input1 (graph 0)
     amix (graph 0) -> Stream #0:0 (libvorbis)
     Stream #2:0 -> #0:1 (rawvideo -> libx264)
    Press [q] to stop, [?] for help
    ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred3.22 bitrate=10423.3kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred25.25 bitrate=11011.0kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred5.76 bitrate=11013.7kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred27.25 bitrate=11175.4kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred7.76 bitrate=11168.7kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred8.24 bitrate=11176.4kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred55.48 bitrate=11243.8kbits/s    
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
    ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
    frame=12871 fps= 30 q=-1.0 Lsize=  542369kB time=00:07:09.31 bitrate=10349.3kbits/s    
    video:539762kB audio:2363kB subtitle:0 global headers:3kB muxing overhead 0.044476%
    [libx264 @ 0x21ae560] frame I:52    Avg QP:15.46  size:725888
    [libx264 @ 0x21ae560] frame P:12819 Avg QP:18.26  size: 40172
    [libx264 @ 0x21ae560] mb I  I16..4: 100.0%  0.0%  0.0%
    [libx264 @ 0x21ae560] mb P  I16..4:  2.6%  0.0%  0.0%  P16..4: 18.1%  0.0%  0.0%  0.0%  0.0%    skip:79.3%
    [libx264 @ 0x21ae560] coded y,uvDC,uvAC intra: 57.8% 49.8% 25.3% inter: 8.9% 8.7% 2.2%
    [libx264 @ 0x21ae560] i16 v,h,dc,p: 23% 29% 32% 16%
    [libx264 @ 0x21ae560] i8c dc,h,v,p: 45% 28% 18%  9%
    [libx264 @ 0x21ae560] kb/s:10306.26

    Please help me, I am really close to get this working !

    UPDATE : The desktop audio is out of sync when skipping filter_complex and microphone also, bit in a smaller amount. Using copy instead of libvorbis does not change anything either.