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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.

• How can i preview FFmpeg audio/video modifications without processing an output in android application ? [FFMPEG, android]

  12 février 2019, par akash varlani

  Basically I am developing a video editing app that lets user choose some of their images and create video slide show with music.

  I am using FFMPEG to generate video slide show from images but the problem is I can only show preview of the video after executing FFMPEG command.

  Is googled so many blogs and all and I know there is a possible way available to display preview of the output.

  Check below image from reference app. I am developing something similar to this app. There is an option to replace the image in video. This app displays instant preview once I change the image.

  

  Here is the link of the reference app if anyone wants to check :
  https://play.google.com/store/apps/details?id=com.newbiz.mvmaster

  Clicking an EXPORT button this app lets you generate video output. I can do that as I know FFMPEG and how to generate OUTPUT FILE using FFMPEG but what I don’t know is how to display quick preview of OUTPUT VIDEO before generating actual VIDEO FILE.

  On my UBUNTU device I can view output of FFMPEG command using FFPLAY tool but how to do the same on android device.

  Some useful link :
  http://androidwarzone.blogspot.com/2011/12/ffmpeg4android.html