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• Error -138 returns "Error number -138 occurred"

  29 avril 2016, par bot1131357

  I am trying to create a program that listens for a period of time, and then times out so that it can return to work on other tasks and retry again later. Here is the code I am testing with :

  AVFormatContext *pFormatCtx = NULL;
  
  AVCodecContext *codecCtx = NULL;
  
  AVCodec *codec;
  
  int ret = 0;
  
  
  
  // Register all formats and codecs
  
  av_register_all();
  
  avformat_network_init(); // for network streaming
  
  
  
  AVDictionary *d = NULL;           // "create" an empty dictionary
  
  av_dict_set(&d, "timeout", "5", 0); // add an entry
  
  av_dict_set(&d, "rtsp_flags", "listen", 0); // add an entry
  
  
  
  char filename[100];
  
  sprintf_s(filename, sizeof(filename), "%s", "rtsp://127.0.0.1:8554/demo");
  
  
  
  
  
  //:::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::
  
  printf_s("Open video file.\n");
  
  // Open video file
  
  ret = avformat_open_input(&pFormatCtx, filename, NULL, &d);   // Returns -138 here 
  
  if (ret <0)
  
  {
  
      printf_s("Failed: cannot open input.\n");
  
      av_strerror(ret, errbuf, ERRBUFFLEN);
  
      fprintf(stderr, "avformat_open_input() fail: %s\n", errbuf);
  
      continue;
  
      //return -1; // Couldn't find stream information
  
  }

  In the listening mode, avformat_open_input() returns -138. Using av_strerror() gives the following explanation : "Error number -138 occurred"

  Is this an Easter egg ? What does -138 stand for ?

• 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.

• Ffmpeg command to efficiently merge mp3 audio file and jpeg image file to a video file

  9 juin 2020, par Eugen

  I'm looking for an efficient way to merge a large .mp3 file and a single .jpg image to a video format for youtube. If I use :

  



  ffmpeg -loop 1 -i image.jpg -i audio.mp3 -c:v libx264 -c:a aac -strict experimental -b:a 192k -shortest output.mp4


  



  the result is very slow, especially for large .mp3 files. Is it possible to do it more efficiently with ffmpeg ? I know there must be a way to do it much more efficiently, like all the online converters are doing it.