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  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • 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 (...)

  • Changer son thème graphique

    22 février 2011, par

    Le thème graphique ne touche pas à la disposition à proprement dite des éléments dans la page. Il ne fait que modifier l’apparence des éléments.
    Le placement peut être modifié effectivement, mais cette modification n’est que visuelle et non pas au niveau de la représentation sémantique de la page.
    Modifier le thème graphique utilisé
    Pour modifier le thème graphique utilisé, il est nécessaire que le plugin zen-garden soit activé sur le site.
    Il suffit ensuite de se rendre dans l’espace de configuration du (...)

Sur d’autres sites (9291)

  • combine multiple mp4 videos and images

    8 juin 2018, par wdsfds

    so I have a folder of images, 1/20 named *.png and a folder of mp4’s named *.mp4.

    I want to create a video in this order :

    • 1.png for 3 sec
    • 1.mp4
    • 2.png for 3 sec
    • 3.mp4
    • etc

    Is there a way I can display each png for 3 seconds and then show the respective mp4 using ffmpeg ? I know I can convert each picture to a 3 second video invididually using this command and the framerate differences will be a problem (1/3 vs 60), but I’m not very experienced with command line video editing :

    ffmpeg -r 1/3 -i 1.png -vcodec mpeg4 1_intro.mp4

    ffprobe output :

    ffprobe version 4.0 Copyright (c) 2007-2018 the FFmpeg developers
    
  built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
    
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
    
  libavutil      56. 14.100 / 56. 14.100
    
  libavcodec     58. 18.100 / 58. 18.100
    
  libavformat    58. 12.100 / 58. 12.100
    
  libavdevice    58.  3.100 / 58.  3.100
    
  libavfilter     7. 16.100 /  7. 16.100
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  1.100 /  5.  1.100
    
  libswresample   3.  1.100 /  3.  1.100
    
  libpostproc    55.  1.100 / 55.  1.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.71.100
    
  Duration: 00:00:32.69, start: 0.000000, bitrate: 7039 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt709/unknown/unknown), 1920x1080 [SAR 1:1 DAR 16:9], 7004 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 43 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    output of out.mp4

    ffprobe version 4.0 Copyright (c) 2007-2018 the FFmpeg developers
    
  built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
    
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
    
  libavutil      56. 14.100 / 56. 14.100
    
  libavcodec     58. 18.100 / 58. 18.100
    
  libavformat    58. 12.100 / 58. 12.100
    
  libavdevice    58.  3.100 / 58.  3.100
    
  libavfilter     7. 16.100 /  7. 16.100
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  1.100 /  5.  1.100
    
  libswresample   3.  1.100 /  3.  1.100
    
  libpostproc    55.  1.100 / 55.  1.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:10:19.13, start: 0.000000, bitrate: 5689 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 2560x1440 [SAR 1:1 DAR 16:9], 5565 kb/s, 53.91 fps, 60 tbr, 90k tbn, 120 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 114 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

  • How to use Google's Cloud Speech-to-Text API to transcribe a video using the REST API

    8 juin 2018, par mrb

    I’d like to have the transcript of 2 people speaking in a video, but I get an empty response from the Cloud Speech-to-Text API

    Approach :

    I have a 56 minute video file containing a conversation between two people. I would like to have the transcript of that conversation, and I would like to use Google’s Cloud Speech-to-Text API to get that.

    To save a little on my Google Cloud Storage I converted to video to audio first by using mmpeg.

    First I’d tried to figure out the audio codec by using the command below, and it looks like AAC.
    ffmpeg -i video.mp4

    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'videoplayback.mp4':
    
  Metadata:
    
    major_brand     : mp42
    
    minor_version   : 0
    
    compatible_brands: isommp42
    
    creation_time   : 2015-12-30T08:17:14.000000Z
    
  Duration: 00:56:03.99, start: 0.000000, bitrate: 362 kb/s
    
    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 490x360 [SAR 1:1 DAR 49:36], 264 kb/s,     29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s (default)
    
    Metadata:
    
      creation_time   : 2015-12-30T08:17:31.000000Z
    
      handler_name    : IsoMedia File Produced by Google, 5-11-2011    

    So I took that from the video by using :
    ffmpeg -i video.mp4 -vn -acodec copy myaudio.aac

    Details so far :
    ffmpeg -i myaudio.aac
    Outputs :

    Input #0, aac, from 'myaudio.aac':
    
  Duration: 00:56:47.49, bitrate: 97 kb/s
    
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 97 kb/s

    After that I converted it to opus because I’m told that opus is better
    ffmpeg -i myaudio.aac -acodec libopus -b:a 97k -vbr on -compression_level 10 myaudio.opus

    Info so far :
    opusinfo myaudio.opus

    User comments section follows...
    
    encoder=Lavc58.18.100 libopus
    
Opus stream 1:
    
    Pre-skip: 312
    
    Playback gain: 0 dB
    
    Channels: 2
    
    Original sample rate: 48000Hz
    
    Packet duration:   20.0ms (max),   20.0ms (avg),   20.0ms (min)
    
    Page duration:   1000.0ms (max), 1000.0ms (avg), 1000.0ms (min)
    
    Total data length: 29956714 bytes (overhead: 0.872%)
    
    Playback length: 56m:03.990s
    
    Average bitrate: 71.24 kb/s, w/o overhead: 70.62 kb/s

    I this point I uploaded the myaudio.opus to the Google Cloud Storage.

    curl POST 1
    I started the speech recognition by doing a POST with curl :

    curl --request POST  --header "Content-Type: application/json" --url 'https://speech.googleapis.com/v1/speech:longrunningrecognize?fields=done%2Cerror%2Cmetadata%2Cname%2Cresponse&key={MY_API_KEY}' --data '{"audio": {"uri": "gs://{MY_BUCKET}/myaudio.opus"},"config": {"encoding": "OGG_OPUS", "sampleRateHertz": 48000, "languageCode": "en-US"}}'

    Response : {"name": "123456789"}
    123456789 was not the actual value.

    curl GET 1
    Now I wanted to have the results :

    curl --request GET --url 'https://speech.googleapis.com/v1/operations/123456789?fields=done%2Cerror%2Cmetadata%2Cname%2Cresponse&key={MY_API_KEY}'

    This gave me the error : Error : Unable to recognize speech, possible error in encoding or channel config. Please correct the config and retry the request.

    So I updated the encoding configuration from OGG_OPUS to LINEAR16.

    curl POST 2
    Did the post again :

    curl --request POST  --header "Content-Type: application/json" --url 'https://speech.googleapis.com/v1/speech:longrunningrecognize?fields=done%2Cerror%2Cmetadata%2Cname%2Cresponse&key={MY_API_KEY}' --data '{"audio": {"uri": "gs://{MY_BUCKET}/myaudio.opus"},"config": {"encoding": "LINEAR16", "sampleRateHertz": 48000, "languageCode": "en-US"}}'

    Response : {"name": "987654321"}

    curl GET 2

    curl --request GET --url 'https://speech.googleapis.com/v1/operations/987654321?fields=done%2Cerror%2Cmetadata%2Cname%2Cresponse&key={MY_API_KEY}'

    Response :

    {
    
  "name": "987654321",
    
  "metadata": {
    
    "@type": "type.googleapis.com/google.cloud.speech.v1.LongRunningRecognizeMetadata",
    
    "progressPercent": 100,
    
    "startTime": "2018-06-08T11:01:24.596504Z",
    
    "lastUpdateTime": "2018-06-08T11:01:51.825882Z"
    
  },
    
  "done": true
    
}

    The problem is that I don’t get the actual transcription. According the the documentation there should be a response key in the response containing the data.

    Since I’m kinda stuck here I’d like to know if I’m doing something completely wrong. I don’t have any technical or resource limitation so all suggestions are very welcome ! Also happy to change my approach.

    Thanks in advance ! Cheers

  • FFMPEG MKV Causing Errors in DASH JS

    24 juin 2018, par Mike

    I’m getting the following browser errors (on all browsers) using Dash JS when transcoding and MKV file :

    ERROR DOMException: Failed to read the 'buffered' property from 'SourceBuffer': This SourceBuffer has been removed from the parent media source.

    and...

    dash.all.min.js:26 Uncaught (in promise) DOMException: Failed to load because no supported source was found.

    What’s weird is I have no issues when I transcode a MP4 file. I’m using FFMPEG in conjunction with Bento4 to build MPEG DASH and HLS files for my video player.

    What I did to single out FFMPEG was to transcode a video that gave me errors on my test server on my local machine (which works) and start the Bento4 process on that file. Doing that, I had no issues and everything played just fine.

    I have removed FFMPEG and reinstalled it multiple times and I always get the same result. I’m sure I screwed something up on my server, but for the life of me I can’t seem to figure out where to start with fixing the issue.

    FFMPEG Version

    ffmpeg version N-91321-ge85c608 Copyright (c) 2000-2018 the FFmpeg developers
    

    
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
    

    
configuration: 
    
--prefix=/root/ffmpeg_build 
    
--pkg-config-flags=--static 
    
--extra-cflags=-I/root/ffmpeg_build/include 
    
--extra-ldflags=-L/root/ffmpeg_build/lib 
    
--extra-libs=-lpthread 
    
--extra-libs=-lm 
    
--bindir=/root/bin 
    
--enable-gpl 
    
--enable-libfdk_aac 
    
--enable-libfreetype 
    
--enable-libmp3lame 
    
--enable-libopus 
    
--enable-libvorbis 
    
--enable-libtheora 
    
--enable-libx264 
    
--enable-nonfree
    
libavutil      56. 18.102 / 56. 18.102
    
libavcodec     58. 20.102 / 58. 20.102
    
libavformat    58. 17.100 / 58. 17.100
    
libavdevice    58.  4.101 / 58.  4.101
    
libavfilter     7. 25.100 /  7. 25.100
    
libswscale      5.  2.100 /  5.  2.100
    
libswresample   3.  2.100 /  3.  2.100
    
libpostproc    55.  2.100 / 55.  2.100

    FFMPEG Command

    ffmpeg 
    
-i ${DIRECTORY}/${INPUT_FILE} 
    
-progress ${DIRECTORY}/transcode.log 
    
-s 1920x1080 
    
-c:v libx264 
    
-b:v 3000k 
    
-c:a aac 
    
-b:a 32k 
    
-minrate 3000k 
    
-maxrate 3000k 
    
-bufsize 6000k 
    
-g 96 
    
-keyint_min 96 
    
-sc_threshold 0 
    
-profile:v high 
    
-flags +cgop 
    
-movflags faststart 
    
-preset ultrafast 
    
-pix_fmt yuv420p 
    
${DIRECTORY}/ffmpeg_1920_1080_3000.mp4 &> ${DIRECTORY}/ffmpeg.log

    Also, I get no errors and and if I access the output files directly, they play just fine.

    I’m sure I’m not including all the information needed to troubleshoot this, so let me know if there is better information I can provide.

    What would cause FFMPEG to transcode MP4 and not MKV ?

    EDIT
    One last thing, I converted the MKV to an MP4 then used the above command and it worked. It’s like MP4 to MP4 is fine, but MKV to MP4 is broke.

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