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  • Modifier la date de publication

    21 juin 2013, par

    Comment changer la date de publication d’un média ?
    Il faut au préalable rajouter un champ "Date de publication" dans le masque de formulaire adéquat :
    Administrer > Configuration des masques de formulaires > Sélectionner "Un média"
    Dans la rubrique "Champs à ajouter, cocher "Date de publication "
    Cliquer en bas de la page sur Enregistrer

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

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  • Convert MPEG4 to MPEGTS on Android with FFmpeg

    3 juin 2013, par Ardoramor

    Ok, so obviously I know very little to none about ffmpeg API when I made the original post... it is quite overwhelming when one starts learning about digital media and conversion details. After reading quite a bit more and going through ffmpeg source, I was able to get a working output from mp4 to mpegts. The concept is similar to executing :

    ffmpeg -i in.mp4 -vcodec copy -acodec copy -vbsf h264_mp4toannexb out.ts

    But as I mentioned before, I need to implement it with ffmpeg API in C.

    So, although I am able to generate a playable .ts file, its video and audio streams are not synced. That is, playing them back on Android tablet plays the video very slowly while audio is playing at normal speed and then (once audio stream ends) video plays at normal speed to the end. Playing the same generated .ts file in VLC produces a very condensed audio (as though fast-forwarded) and plays video fine.

    There are still many aspects of media conversion that I am not familiar with. I am sure that some of them prevent me from successful conversion.

    Here is some information (via ffprobe) about the files :
    - in.mp4 - file generated via Android recording - MPEG4 (H.264 + AAC)
    - ffmpeg.ts - file generated via ffmpeg conversion - MPEG2TS (H.264 + AAC)
    - out.ts - file generated via my code - MPEGTS (H.264 + AAC)

    in.mp4

    filename=in.mp4
    
nb_streams=2
    
format_name=mov,mp4,m4a,3gp,3g2,mj2
    
format_long_name=QuickTime/MPEG-4/Motion JPEG 2000 format
    
start_time=0:00:00.000000
    
duration=0:00:09.961383
    
size=4.730 Mibyte
    
bit_rate=3.983 Mbit/s
    
TAG:major_brand=isom
    
TAG:minor_version=0
    
TAG:compatible_brands=isom3gp4
    
TAG:creation_time=2013-05-28 17:06:57

    ffmpeg.ts

    filename=ffmpeg.ts
    
nb_streams=2
    
format_name=mpegts
    
format_long_name=MPEG-2 transport stream format
    
start_time=0:00:01.400000
    
duration=0:00:09.741267
    
size=5.132 Mibyte
    
bit_rate=4.419 Mbit/s

    out.ts

    filename=out.ts
    
nb_streams=2
    
format_name=mpegts
    
format_long_name=MPEG-2 transport stream format
    
start_time=0:00:00.000000
    
duration=0:00:09.741267
    
size=5.166 Mibyte
    
bit_rate=4.449 Mbit/s

    Firstly, I was unable to affect my output file's start_time. Next, upon examining the -show_packets output of probe, I saw the following :

    ffmpeg.ts

    [PACKET]
    
codec_type=video
    
stream_index=0
    
pts=N/A
    
pts_time=N/A
    
dts=N/A
    
dts_time=N/A
    
duration=0
    
duration_time=0:00:00.000000
    
size=20.506 Kibyte
    
pos=564
    
flags=K
    
[/PACKET]
    
[PACKET]
    
codec_type=video
    
stream_index=0
    
pts=N/A
    
pts_time=N/A
    
dts=N/A
    
dts_time=N/A
    
duration=0
    
duration_time=0:00:00.000000
    
size=11.727 Kibyte
    
pos=22936
    
flags=_
    
[/PACKET]
    
...
    
[PACKET]
    
codec_type=audio
    
stream_index=1
    
pts=126000
    
pts_time=0:00:01.400000
    
dts=126000
    
dts_time=0:00:01.400000
    
duration=2089
    
duration_time=0:00:00.023211
    
size=285.000 byte
    
pos=109416
    
flags=K
    
[/PACKET]
    
[PACKET]
    
codec_type=audio
    
stream_index=1
    
pts=128089
    
pts_time=0:00:01.423211
    
dts=128089
    
dts_time=0:00:01.423211
    
duration=2089
    
duration_time=0:00:00.023211
    
size=374.000 byte
    
pos=-1
    
flags=K
    
[/PACKET]
    
...
    
[PACKET]
    
codec_type=video
    
stream_index=0
    
pts=N/A
    
pts_time=N/A
    
dts=N/A
    
dts_time=N/A
    
duration=0
    
duration_time=0:00:00.000000
    
size=20.000 Kibyte
    
pos=87232
    
flags=_
    
[/PACKET]
    
[PACKET]
    
codec_type=video
    
stream_index=0
    
pts=N/A
    
pts_time=N/A
    
dts=N/A
    
dts_time=N/A
    
duration=0
    
duration_time=0:00:00.000000
    
size=16.852 Kibyte
    
pos=112800
    
flags=_
    
[/PACKET]

    out.ts

    [PACKET]
    
codec_type=audio
    
stream_index=1
    
pts=0
    
pts_time=0:00:00.000000
    
dts=0
    
dts_time=0:00:00.000000
    
duration=2089
    
duration_time=0:00:00.023211
    
size=285.000 byte
    
pos=22936
    
flags=K
    
[/PACKET]
    
[PACKET]
    
codec_type=audio
    
stream_index=1
    
pts=1024
    
pts_time=0:00:00.011378
    
dts=1024
    
dts_time=0:00:00.011378
    
duration=2089
    
duration_time=0:00:00.023211
    
size=374.000 byte
    
pos=23312
    
flags=K
    
[/PACKET]
    
...
    

    
[PACKET]
    
codec_type=video
    
stream_index=0
    
pts=N/A
    
pts_time=N/A
    
dts=N/A
    
dts_time=N/A
    
duration=0
    
duration_time=0:00:00.000000
    
size=11.727 Kibyte
    
pos=25004
    
flags=_
    
[/PACKET]
    
[PACKET]
    
codec_type=audio
    
stream_index=1
    
pts=7168
    
pts_time=0:00:00.079644
    
dts=7168
    
dts_time=0:00:00.079644
    
duration=2089
    
duration_time=0:00:00.023211
    
size=299.000 byte
    
pos=55460
    
flags=K
    
[/PACKET]

    As you can see, ffmpeg.ts starts out with video packets that do not have pts/dts. The audio packets that follow contain pts/dts. This repeats until the end. All video packets do not have pts/dts according to ffprobe output.

    However, out.ts starts with audio packets and alternate with video packets. Here, video packets also do not have pts/dts. The difference is that here there is one video packet between a series of audio packets. What happened to the rest of the video packets (ffmpeg.ts has 5 audio followed by 5 video).

    Obviously, I'm still learning and don't know way too much yet... Does anything jump out as obvious a problem to anyone ? I will greatly appreciate any info / suggestions but will continue to grind at it !!

  • Ffmpeg send duration of video to client (using node-fluent-ffmpeg)

    26 mai 2013, par Vprnl

    I'm really new to the world of ffmpeg so please excuses me if this is a stupid queston.

    I'm using the module Node-fluent-ffmpeg to stream a movie and convert it from avi to webm with FFMPEG.

    So far so good (it plays the video), but I'm having trouble parsing the duration to the player. It also gives me an error even though I plays the video.

    my code is as followed :

    var stat = fs.statSync(movie);
    

    
var start = 0;
    
var end = 0;
    
var range = req.header('Range');
    
if (range != null) {
    
start = parseInt(range.slice(range.indexOf('bytes=')+6,
    
  range.indexOf('-')));
    
end = parseInt(range.slice(range.indexOf('-')+1,
    
  range.length));
    
}
    
if (isNaN(end) || end == 0) end = stat.size-1;
    
if (start > end) return;
    

    
var duration = (end / 1024) * 8 / 1024;
    

    
res.writeHead(206, { // NOTE: a partial http response
    
    'Connection':'close',
    
    'Content-Type':'video/webm',
    
    'Content-Length':end - start,
    
    'Content-Range':'bytes '+start+'-'+end+'/'+stat.size,
    
    'Transfer-Encoding':'chunked'
    
});
    

    
var proc = new ffmpeg({ source: movie, nolog: true, priority: 1, timeout:15000})
    
    .toFormat('webm')
    
    .addOptions(['-probesize 900000', '-analyzeduration 0', '-minrate 1024k', '-maxrate 1024k', '-bufsize 1835k', '-t '+duration+' -ss'])
    
    .writeToStream(res, function(retcode, error){
    
    if (!error){
    
        console.log('file has been converted succesfully',retcode);
    
    }else{
    
        console.log('file conversion error',error);
    
    }
    
});

    I set the header with a start and a end based on this article : http://delog.wordpress.com/2011/04/25/stream-webm-file-to-chrome-using-node-js/

    I calculate the length in seconds in the variable duration.

    The error FFmpeg is giving me is :

        file conversion error ffmpeg version N-52458-gaa96439 Copyright (c) 2000-2013 the FFmpeg developers
    
      built on Apr 24 2013 22:19:32 with gcc 4.8.0 (GCC)
    
      configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --e
    
    nable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable
    
    -libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --ena
    
    ble-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwola
    
    me --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enabl
    
    e-libxvid --enable-zlib
    
      libavutil      52. 27.101 / 52. 27.101
    
      libavcodec     55.  6.100 / 55.  6.100
    
      libavformat    55.  3.100 / 55.  3.100
    
      libavdevice    55.  0.100 / 55.  0.100
    
      libavfilter     3. 60.101 /  3. 60.101
    
      libswscale      2.  2.100 /  2.  2.100
    
      libswresample   0. 17.102 /  0. 17.102
    
      libpostproc    52.  3.100 / 52.  3.100
    
    Input #0, avi, from 'C:/temp/test.avi':
    
      Metadata:
    
        encoder         : Nandub v1.0rc2
    
      Duration: 00:01:09.78, start: 0.000000, bitrate: 1517 kb/s
    
        Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 640x352, 23.98 tbr, 23.98 tbn, 23.98 tbc
    
        Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 222 kb/s
    
    [libvpx @ 0036db20] v1.2.0
    
    Output #0, webm, to 'pipe:1':
    
      Metadata:
    
        encoder         : Lavf55.3.100
    
        Stream #0:0: Video: vp8, yuv420p, 640x352, q=-1--1, 200 kb/s, 1k tbn, 23.98 tbc
    
        Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp
    
    Stream mapping:
    
      Stream #0:0 -> #0:0 (msmpeg4 -> libvpx)
    
      Stream #0:1 -> #0:1 (mp3 -> libvorbis)

    The client side player (which is VideoJs) says the file is infinite/NaN in length.

    I feel like I'm pretty close to a solution but my inexperience with the subject matter prohibits me from getting it to work. If I'm unclear in any way please let me know. (I have a tendency of explaining things fuzzy.)

    Thanks in advance !

    [EDIT]

    I removed the duration bit because it has nothing to do with the issue. I checked the response header of the client and saw :

     Accept-Ranges:bytes
    
 Connection:keep-alive
    
 Content-Length:13232127
    
 Content-Range:bytes 0-13232127/13232128
    
 Content-Type:video/webm

    Why can't the client figure out the duration even though it receives it in the header ?

  • RTMP to HLS : Segmentation fault

    25 mai 2013, par Wildan Muhlis

    I have tried convert RTMP to HLS with ffmpeg. The segments created without -segment_format mpegts parameter, but it resulting corrupted .m3u8 file, If I add the parameter, segments couldn't created due to below error message.

    Anyone know how to solve it ?

    ffmpeg encoding profile :

    ffmpeg -y -i ${input}" live=1 swfVfy=1" \
    
-ar 48000 \
    
-ab 64k \
    
-s ${WIDTH}x${HEIGHT} \
    
-vcodec libx264 \
    
-b:v ${BR} \
    
-partitions +parti4x4+partp8x8+partb8x8 \
    
-subq 7 \
    
-trellis 0 \
    
-refs 0 \
    
-coder 0 \
    
-me_range 16 \
    
-keyint_min 25 \
    
-sc_threshold 40 \
    
-i_qfactor 0.71 \
    
-bt 200k \
    
-maxrate ${BR} \
    
-bufsize ${BR} \
    
-rc_eq 'blurCplx^(1-qComp)' \
    
-qcomp 0.6 \
    
-qmin 30 \
    
-qmax 51 \
    
-qdiff 4 \
    
-level 30 \
    
-aspect ${WIDTH}:${HEIGHT} \
    
-g 30 \
    
-async 2 \
    
-flags -global_header -map 0 \
    
-f segment \
    
-flags +loop \
    
-segment_time 10 \
    
-segment_list ${SEGMENT_LIST} \
    
-segment_format mpegts\
    
-segment_list_flags +live \
    
${OUTPUT}

    Error log :

    ffmpeg version N-53211-g5918b7a Copyright (c) 2000-2013 the FFmpeg developers
    
  built on May 18 2013 09:16:55 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
    
  configuration: --as=yasm --enable-gpl --enable-pthreads --disable-ffserver --disable-shared --enable-static --enable-gpl --enable-libfdk_aac --enable-libmp3lame --enable-libtheora --enable-libvpx --enable-libx264 --enable-librtmp --enable-nonfree
    
  libavutil      52. 33.100 / 52. 33.100
    
  libavcodec     55. 10.100 / 55. 10.100
    
  libavformat    55.  7.100 / 55.  7.100
    
  libavdevice    55.  0.100 / 55.  0.100
    
  libavfilter     3. 68.101 /  3. 68.101
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
rtmp server sent error
    
Metadata:
    
  audiocodecid          .mp3
    
  videokeyframe_frequency5.00
    
  avclevel              30.00
    
  videodevice           ASUS USB2.0 Webcam
    
  audiosamplerate       44100.00
    
  audiochannels         2.00
    
  width                 320.00
    
  videodatarate         200.00
    
  presetname            Custom
    
  audioinputvolume      75.00
    
  creationdate          Sat May 25 20:02:26 2013
    
  videocodecid          avc1
    
  audiodevice           Microphone (Realtek High Defini
    
  avcprofile            66.00
    
  audiodatarate         96.00
    
  height                240.00
    
  framerate             30.00
    
[flv @ 0x2df2960] max_analyze_duration 5000000 reached at 5018000 microseconds
    
[flv @ 0x2df2960] decoding for stream 0 failed
    
Input #0, flv, from 'rtmp://122.22.117.60:1935/oflaDemo/livestream live=1 swfVfy=1':
    
  Metadata:
    
    videokeyframe_frequency: 5
    
    avclevel        : 30
    
    videodevice     : ASUS USB2.0 Webcam
    
    keywords        : 
    
    audiochannels   : 2
    
    presetname      : Custom
    
    copyright       : 
    
    audioinputvolume: 75
    
    creationdate    : Sat May 25 20:02:26 2013
    
                    : 
    
    author          : 
    
    audiodevice     : Microphone (Realtek High Defini
    
    avcprofile      : 66
    
    title           : 
    
    description     : 
    
    rating          : 
    
  Duration: N/A, start: 0.000000, bitrate: 303 kb/s
    
    Stream #0:0: Video: h264 (Baseline), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 204 kb/s, 30 tbr, 1k tbn, 60 tbc
    
    Stream #0:1: Audio: mp3, 44100 Hz, stereo, s16p, 98 kb/s
    
Segmentation fault

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