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  • MediaSPIP version 0.1 Beta

    16 avril 2011, par

    MediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
    Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
    Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
    Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)

  • 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 (...)

  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

Sur d’autres sites (10203)

  • Google Speech - Streaming Request Returns EOF Error

    16 octobre 2017, par Josh

    Using Go, I’m taking a RTMP stream, transcoding it to FLAC (using ffmpeg) and attempting to stream to Google’s Speech API to transcribe the audio. However, I keep getting EOF errors when sending the data. I can’t find any information on this error in the docs so I’m not exactly sure what’s causing it.

    I’m chunking the received data into 3s clips (length isn’t relevant as long as it’s less than the maximum length of a streaming recognition request).

    Here is the core of my code :

    func main() {

       done := make(chan os.Signal)
       received := make(chan []byte)

       go receive(received)
       go transcribe(received)

       signal.Notify(done, os.Interrupt, syscall.SIGTERM)

       select {
       case <-done:
           os.Exit(0)
       }
    }

    func receive(received chan<- []byte) {
       var b bytes.Buffer
       stdout := bufio.NewWriter(&b)

       cmd := exec.Command("ffmpeg", "-i", "rtmp://127.0.0.1:1935/live/key", "-f", "flac", "-ar", "16000", "-")
       cmd.Stdout = stdout

       if err := cmd.Start(); err != nil {
           log.Fatal(err)
       }

       duration, _ := time.ParseDuration("3s")
       ticker := time.NewTicker(duration)

       for {
           select {
           case <-ticker.C:
               stdout.Flush()
               log.Printf("Received %d bytes", b.Len())
               received <- b.Bytes()
               b.Reset()
           }
       }
    }

    func transcribe(received <-chan []byte) {
       ctx := context.TODO()

       client, err := speech.NewClient(ctx)
       if err != nil {
           log.Fatal(err)
       }

       stream, err := client.StreamingRecognize(ctx)
       if err != nil {
           log.Fatal(err)
       }

       // Send the initial configuration message.
       if err = stream.Send(&speechpb.StreamingRecognizeRequest{
           StreamingRequest: &speechpb.StreamingRecognizeRequest_StreamingConfig{
               StreamingConfig: &speechpb.StreamingRecognitionConfig{
                   Config: &speechpb.RecognitionConfig{
                       Encoding:        speechpb.RecognitionConfig_FLAC,
                       LanguageCode:    "en-GB",
                       SampleRateHertz: 16000,
                   },
               },
           },
       }); err != nil {
           log.Fatal(err)
       }

       for {
           select {
           case data := <-received:
               if len(data) > 0 {
                   log.Printf("Sending %d bytes", len(data))
                   if err := stream.Send(&speechpb.StreamingRecognizeRequest{
                       StreamingRequest: &speechpb.StreamingRecognizeRequest_AudioContent{
                           AudioContent: data,
                       },
                   }); err != nil {
                       log.Printf("Could not send audio: %v", err)
                   }
               }
           }
       }
    }

    Running this code gives this output :

    2017/10/09 16:05:00 Received 191704 bytes
    2017/10/09 16:05:00 Saving 191704 bytes
    2017/10/09 16:05:00 Sending 191704 bytes
    2017/10/09 16:05:00 Could not send audio: EOF

    2017/10/09 16:05:03 Received 193192 bytes
    2017/10/09 16:05:03 Saving 193192 bytes
    2017/10/09 16:05:03 Sending 193192 bytes
    2017/10/09 16:05:03 Could not send audio: EOF

    2017/10/09 16:05:06 Received 193188 bytes
    2017/10/09 16:05:06 Saving 193188 bytes
    2017/10/09 16:05:06 Sending 193188 bytes // Notice that this doesn't error

    2017/10/09 16:05:09 Received 191704 bytes
    2017/10/09 16:05:09 Saving 191704 bytes
    2017/10/09 16:05:09 Sending 191704 bytes
    2017/10/09 16:05:09 Could not send audio: EOF

    Notice that not all of the Sends fail.

    Could anyone point me in the right direction here ? Is it something to do with the FLAC headers or something ? I also wonder if maybe resetting the buffer causes some of the data to be dropped (i.e. it’s a non-trivial operation that actually takes some time to complete) and it doesn’t like this missing information ?

    Any help would be really appreciated.

  • Revision c88f1ec8ca : Android.mk file for vpx unittests These changes are to support automated regres

    11 novembre 2013, par Joshua Litt

    Changed Paths :
     Add /test/android/Android.mk


     Add /test/android/README


     Add /test/android/get_files.py



    Android.mk file for vpx unittests

    These changes are to support automated regressions of vpx on android
    new file : test/android/Android.mk
    new file : test/android/README
    new file : test/android/get_files.py

    Change-Id : I52c8e9daf3676a3561badbe710ec3a16fed72abd

  • how to render android screen to a web page in realtime

    14 août 2012, par Rain Lee

    i'm working on something to remote control android from web browser. something like canvas or flash will draw the android device's screen, and i want it to be very smoothly.

    as i know, we can capture android screen using frame buffer. there a lot of articles about this. But, the fps is very lot, 4 5 fbs at most. is there any way to improve ?

    will something like ffmpeg help?