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  • 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 (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (8737)

  • ffmpeg cannot open a simple microsoft wav file exported with Audacity

    18 février 2014, par sebpiq

    I have exported a sound file to microsoft wav using Audacity.
    I am trying to open this file with ffmpeg :

    ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg

    and here's the ouput I get :

    fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
    
  configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
    
  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
    
  libpostproc    52.  2.100 / 52.  2.100
    
[dca @ 0x7fd30c013600] Not a valid DCA frame
    

    
... SNIP ...
    

    
[dca @ 0x7fd5bc013600] Invalid bit allocation index
    
[dca @ 0x7fd5bc013600] error decoding block
    
    Last message repeated 3 times
    
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
    
[dca @ 0x7fd5bc013600] error decoding block
    
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
    
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
    
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
steps-stereo-16b-44khz.wav: could not find codec parameters

    If I export the same file to .ogg or .aiff, no problem, the following works fine :

    ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg

    Any idea what could be wrong ?

    A link to my wav file so you can try to reproduce.

    NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

    EDIT

    Getting file info with another program like sox, works well :

    sox --info steps-stereo-16b-44khz.wav
    

    
Input File     : 'steps-stereo-16b-44khz.wav'
    
Channels       : 2
    
Sample Rate    : 44100
    
Precision      : 16-bit
    
Duration       : 00:00:02.10 = 92608 samples = 157.497 CDDA sectors
    
File Size      : 370k
    
Bit Rate       : 1.41M
    
Sample Encoding: 16-bit Signed Integer PCM

  • ffmpef cannot open a simple microsoft wav file exported with Audacity

    23 juillet 2013, par sebpiq

    I have exported a sound file to microsoft wav using Audacity.
    I am trying to open this file with ffmpeg :

    ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg

    and here's the ouput I get :

    fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
    
  configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
    
  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
    
  libpostproc    52.  2.100 / 52.  2.100
    
[dca @ 0x7fd30c013600] Not a valid DCA frame
    

    
... SNIP ...
    

    
[dca @ 0x7fd5bc013600] Invalid bit allocation index
    
[dca @ 0x7fd5bc013600] error decoding block
    
    Last message repeated 3 times
    
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
    
[dca @ 0x7fd5bc013600] error decoding block
    
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
    
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
    
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
steps-stereo-16b-44khz.wav: could not find codec parameters

    If I export the same file to .ogg or .aiff, no problem, the following works fine :

    ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg

    Any idea what could be wrong ?

    A link to my wav file so you can try to reproduce.

    NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

  • why ffmpeg is not working in new installation

    3 août 2013, par Hitesh Gupta

    I was working on live encoding from FFmpeg from last few days. One day I re-installed my OS and tried to run FFmpeg commands again after configuration. My publish points get in starting but could not started. Why ? Am I missing any configuration required ?

    The command I am trying to run is :

    ffmpeg -y -re -i D:\video2.mp4 -pix_fmt yuv420p -movflags isml+frag_keyframe -f ismv -threads 0 -c:v libx264 -preset fast -profile:v baseline -map 0:v -b:v:0 800k http://localhost/PPS/PublishPoint.isml/Streams(Encode
    
r1)

    Output what I got in command prompt is :

    ffmpeg version N-54772-g53c853e Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jul 16 2013 22:25:42 with gcc 4.7.3 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
    
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
    
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
    
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
    
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
    
enable-libxvid --enable-zlib
    
  libavutil      52. 40.100 / 52. 40.100
    
  libavcodec     55. 18.102 / 55. 18.102
    
  libavformat    55. 12.102 / 55. 12.102
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 81.101 /  3. 81.101
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\video2.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf55.10.100
    
  Duration: 00:00:12.12, start: 0.072562, bitrate: 945 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 704x396 [
    
SAR 1:1 DAR 16:9], 882 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 22050 Hz, stereo, fltp, 64
    
 kb/s
    
    Metadata:
    
      handler_name    : SoundHandler
    
[libx264 @ 00000000047e0860] using SAR=1/1
    
[libx264 @ 00000000047e0860] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
    
[libx264 @ 00000000047e0860] profile Constrained Baseline, level 3.0
    
[libx264 @ 00000000047e0860] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC cod
    
ec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 r
    
ef=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed
    
_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pski
    
p=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 deci
    
mate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyi
    
nt=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=abr mbtree=1
    
 bitrate=800 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1
    
:1.00
    
Output #0, ismv, to 'http://localhost/My_SSMN_PPS/saturday.isml/Streams(Encoder1
    
)':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf55.12.102
    
    Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 7
    
04x396 [SAR 1:1 DAR 16:9], q=-1--1, 800 kb/s, 10000k tbn, 29.97 tbc
    
    Metadata:
    
      handler_name    : VideoHandler
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (h264 -> libx264)
    
Press [q] to stop, [?] for help
    
frame=   13 fps=0.0 q=0.0 size=       2kB time=00:00:00.00 bitrate=N/A dup=2 dro
    
frame=   29 fps= 29 q=0.0 size=       2kB time=00:00:00.00 bitrate=N/A dup=2 dro
    
av_interleaved_write_frame(): Unknown error

    Advanced Thanks.

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