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  • 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 (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • Mediabox : ouvrir les images dans l’espace maximal pour l’utilisateur

    8 février 2011, par

    La visualisation des images est restreinte par la largeur accordée par le design du site (dépendant du thème utilisé). Elles sont donc visibles sous un format réduit. Afin de profiter de l’ensemble de la place disponible sur l’écran de l’utilisateur, il est possible d’ajouter une fonctionnalité d’affichage de l’image dans une boite multimedia apparaissant au dessus du reste du contenu.
    Pour ce faire il est nécessaire d’installer le plugin "Mediabox".
    Configuration de la boite multimédia
    Dès (...)

Sur d’autres sites (5318)

  • Anomalie #3592 : Robots.txt.html et tests de compatibilité mobile de Google

    25 novembre 2019, par b b

    Pas certain que ça soit nécessaire de doublonner en ouvrant de nouveau ce ticket puisque le sujet est discuté par ici https://core.spip.net/issues/4103 :)

  • 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

  • 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

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