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  • 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

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (2778)

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

    24 juillet 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 combining mp4 videos incorrectly

    9 juin 2018, par wdsfds

    i have a list of 20 mp4 files that I need to combine into one video. They are all 60 FPS, but range in size. In between each mp4 file I need to show its ’intro’, a 1.5 second clip about the file.

    Reference :

    • pics/i.png - intro for the nth mp4 file
    • clips/i_intro.mp4 - mp4 intro generated from i.png
    • clips/i.mp4 - original mp4
    • clips/i_resize.mp4 - 720p mp4

    Here is what I’ve done :

    Convert image to mp4 using this command (taken from this post) :

    ffmpeg -y -i pics/{i}.png -f lavfi -i anullsrc -r 60 -pix_fmt yuv420p -ac 2 -ar 44100 -video_track_timescale 90k -t 1.5 clips/{i}_intro.mp4

    Convert i.mp4 to 720p using this command (taken from this post) :

    ffmpeg -i clips/{i}.mp4 -c:v libx264 -crf 23 -preset medium -c:a aac -b:a 128k -vf scale=-2:720,format=yuv420p {i}_resize.mp4

    Created a file like this :

    file 1_intro.mp4
    
file 1_resize.mp4
    
file 2_intro.mp4
    
file 2_resize.mp4
    
...

    Combine all clips :

    ffmpeg -f concat -i {video_list} -c copy out.mp4

    This generates a file, but it is definitely messed up. When I combined the files I got a ton of warning messages like this :

    [mp4 @ 0x7f88fe01de00] Non-monotonous DTS in output stream 0:0;
    
previous: 48966232, current: 8826932; changing to 48966233. This may
    
result in incorrect timestamps in the output file.

    FFprobe for 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: 00:09:35.56, start: 0.000000, bitrate: 4119 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 4221 kb/s, 59.98 fps, 351.56 tbr, 90k tbn, 120 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 122 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    FFprobe for 1.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 '1.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.71.100
    
  Duration: 00:00:40.94, start: 0.000000, bitrate: 7613 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt709/unknown/unknown), 1920x1080 [SAR 1:1 DAR 16:9], 7451 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, 160 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    FFprobe for 1_resized.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 '1_resize.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:32.37, start: 0.000000, bitrate: 3266 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 3127 kb/s, 60 fps, 60 tbr, 15360 tbn, 120 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 129 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    FFprobe for 1_intro.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 '1_intro.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:01.52, start: 0.000000, bitrate: 113 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 4667 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, 2 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    I noticed that the length of the files are different, and the tbn in different, but idk what that means. Thanks for helping !

  • Converting a call center recording to something useful

    9 août 2018, par Abhay

    I have a call center recording (when played it sounds gibberish) for which the mediainfo shows info as

    ion@aurora:~/Inbound$ mediainfo 48401-3405-48403--18042018170000.wav 
    
General
    
Complete name                            : 48401-3405-48403--18042018170000.wav
    
Format                                   : Wave
    
File size                                : 327 KiB
    
Duration                                 : 4mn 11s
    
Overall bit rate                         : 10.7 Kbps
    

    
Audio
    
Format                                   : G.723.1
    
Codec ID                                 : A100
    
Duration                                 : 4mn 11s
    
Bit rate                                 : 10.7 Kbps
    
Channel(s)                               : 2 channels
    
Sampling rate                            : 8 000 Hz
    
Stream size                              : 327 KiB (100%)

    The ffmpeg info shows this as

    ion@aurora:~/Inbound$ ffmpeg -i 48401-3405-48403--18042018170000.wav
    
ffmpeg version N-91330-ga990184 Copyright (c) 2000-2018 the FFmpeg developers
    
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
    
  configuration: --prefix=/home/ion/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/ion/ffmpeg_build/include --extra-ldflags=-L/home/ion/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/ion/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
    
  libavutil      56. 18.102 / 56. 18.102
    
  libavcodec     58. 20.103 / 58. 20.103
    
  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
    
Input #0, wav, from '48401-3405-48403--18042018170000.wav':
    
  Duration: 00:04:11.37, bitrate: 10 kb/s
    
    Stream #0:0: Audio: g723_1 ([0][161][0][0] / 0xA100), 8000 Hz, mono, s16, 10 kb/s
    
At least one output file must be specified

    So I converted this file to PCM using

    ffmpeg -acodec g723_1 -i 48401-3405-48403--18042018170000.wav -acodec pcm_s16le -f wav outnew1.wav

    But the audio still sound gibberish , I tried many variation and only Goldwave worked but that works on windows and with GUI not cli.

    So how can I convert this file to something useful so that atleast I can listen to it , It feels like a challenge now.

    Audio file : https://drive.google.com/open?id=1T54lKaI6IJmOqTPNOA_OkYRz89EQ5F2L

    PS : Use VLC to play audio file

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