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    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
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  • Soumettre bugs et patchs

    10 avril 2011

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Sur d’autres sites (5906)

  • How to configure ffmpeg just to play RTSP videos

    28 février 2018, par Anuran Barman

    I have successfully compiled ffmpeg for android and everything is working fine.

    I have made specific build for each architecture and even with that it’s 9.7-9.9MB in debug version.

    My sole target is just to play RTSP video with authentication.

    What should be the command line options for this while configuring ?

    my current script looks like this

    ./configure \
           --prefix=$prefix \
           --pkg-config=/usr/bin/pkg-config \
           --enable-shared \
           --disable-static \
           --disable-doc \
           --disable-ffmpeg \
           --disable-ffplay \
           --disable-ffprobe \
           --disable-avdevice \
           --disable-symver \
           --cross-prefix=$toolchain/bin/$crossPrefix \
           --target-os=android \
           --arch=arm \
           --enable-cross-compile \
           --sysroot=$sysroot \
           --enable-network \
           --extra-cflags="$mArchFlag" \
           --extra-ldflags="$extraLDFlags"
  • Google Speech API + Go - Transcribing Audio Stream of Unknown Length

    14 février 2018, par Josh

    I have an rtmp stream of a video call and I want to transcribe it. I have created 2 services in Go and I’m getting results but it’s not very accurate and a lot of data seems to get lost.

    Let me explain.

    I have a transcode service, I use ffmpeg to transcode the video to Linear16 audio and place the output bytes onto a PubSub queue for a transcribe service to handle. Obviously there is a limit to the size of the PubSub message, and I want to start transcribing before the end of the video call. So, I chunk the transcoded data into 3 second clips (not fixed length, just seems about right) and put them onto the queue.

    The data is transcoded quite simply :

    var stdout Buffer

    cmd := exec.Command("ffmpeg", "-i", url, "-f", "s16le", "-acodec", "pcm_s16le", "-ar", "16000", "-ac", "1", "-")
    cmd.Stdout = &stdout

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

    ticker := time.NewTicker(3 * time.Second)

    for {
       select {
       case <-ticker.C:
           bytesConverted := stdout.Len()
           log.Infof("Converted %d bytes", bytesConverted)

           // Send the data we converted, even if there are no bytes.
           topic.Publish(ctx, &pubsub.Message{
               Data: stdout.Bytes(),
           })

           stdout.Reset()
       }
    }

    The transcribe service pulls messages from the queue at a rate of 1 every 3 seconds, helping to process the audio data at about the same rate as it’s being created. There are limits on the Speech API stream, it can’t be longer than 60 seconds so I stop the old stream and start a new one every 30 seconds so we never hit the limit, no matter how long the video call lasts for.

    This is how I’m transcribing it :

    stream := prepareNewStream()
    clipLengthTicker := time.NewTicker(30 * time.Second)
    chunkLengthTicker := time.NewTicker(3 * time.Second)

    cctx, cancel := context.WithCancel(context.TODO())
    err := subscription.Receive(cctx, func(ctx context.Context, msg *pubsub.Message) {

       select {
       case <-clipLengthTicker.C:
           log.Infof("Clip length reached.")
           log.Infof("Closing stream and starting over")

           err := stream.CloseSend()
           if err != nil {
               log.Fatalf("Could not close stream: %v", err)
           }

           go getResult(stream)
           stream = prepareNewStream()

       case <-chunkLengthTicker.C:
           log.Infof("Chunk length reached.")

           bytesConverted := len(msg.Data)

           log.Infof("Received %d bytes\n", bytesConverted)

           if bytesConverted > 0 {
               if err := stream.Send(&speechpb.StreamingRecognizeRequest{
                   StreamingRequest: &speechpb.StreamingRecognizeRequest_AudioContent{
                       AudioContent: transcodedChunk.Data,
                   },
               }); err != nil {
                   resp, _ := stream.Recv()
                   log.Errorf("Could not send audio: %v", resp.GetError())
               }
           }

           msg.Ack()
       }
    })

    I think the problem is that my 3 second chunks don’t necessarily line up with starts and end of phrases or sentences so I suspect that the Speech API is a recurrent neural network which has been trained on full sentences rather than individual words. So starting a clip in the middle of a sentence loses some data because it can’t figure out the first few words up to the natural end of a phrase. Also, I lose some data in changing from an old stream to a new stream. There’s some context lost. I guess overlapping clips might help with this.

    I have a couple of questions :

    1) Does this architecture seem appropriate for my constraints (unknown length of audio stream, etc.) ?

    2) What can I do to improve accuracy and minimise lost data ?

    (Note I’ve simplified the examples for readability. Point out if anything doesn’t make sense because I’ve been heavy handed in cutting the examples down.)

  • How do IP camera stream video across home network

    22 janvier 2018, par Ouroboros

    My question is how do the IP camera stream the data from home network to public network. Here’s how I think it can be done :

    1. If I’d to set up something like this using a raspberry pi camera module. I’d probably use port forwarding on my Access Point/Wifi Router. However, clearly, this is not a scalable solution, and there must be something else that off the shelf IP cameras must be doing.

    2. One option is to stream the video (using ffmpeg) to a remove server, and then that remote server can probably again "re-stream" that ? -If this is indeed the case, how is it done ?

    I understand backend architecture very strongly, and have developed fairly complex onces so I do want a fairly technical answer for this one.