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• ffmpeg save hls to m3u8 with mp4 segments

  9 juin 2018, par J. Omen

  I am looking for a command ffmpeg, which saving live input (rtmp or hls) to hls m3u8 with mp4 segments files. I know that it is possible to do, i.e. there is infohttps://bitmovin.com/hls-news-wwdc-2016/ but every command I tries - makes ts files. Anyone know solution ?

• Im getting error when generating thumbnails using ffmpeg in firebase functions

  5 juillet 2020, par Sajjad

  I'm writing this code for generating thumbnail from a video uploaded to firebase storage using FFmpeg, I am saving thumbnails first in the temp directory and then uploading it to firebase storage. The thumbnails are saved in temp dir but I'm getting an error while saving it to firebase storage. (EISDIR ERROR)

  


  const targetTempFilePath = path.join(os.tmpdir(), targetTempFileName);
const targetStorageFilePath = path.join(path.dirname(filePath), targetTempFileName);
ffmpeg.ffprobe(req.body.videoUrl, (err, metaData) => {
    if (err) {
      console.log(err);
    }

    fileDuration = metaData.format.duration;
  });

  ffmpeg(req.body.videoUrl)
    .on("filenames", (filenames) => {
      console.log("Will generate " + filenames.join(", "));
      thumbsFileUrl = "" })
    .on("end", () => {
      console.log(targetTempFilePath);
      admin
        .storage()
        .bucket()
        .upload(targetTempFilePath, {
          destination: targetStorageFilePath,
        })
        .then(() => {
          console.log("Output audio uploaded to", targetStorageFilePath);
          // fs.unlinkSync(tempLocalFile);
          return res.status(200).json({
            success: true,
            thumbsFileUrl: thumbsFileUrl,
            fileDuration: fileDuration,
          });
        })
        .catch((err) => {
          console.log(err);
          return res.json({ success: false });
        });
    })

    .screenshots({
      // Will take screens at 20%, 40%, 60% and 80% of the video
      count: 2,
      folder: targetTempFilePath,
    });


  


• 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.