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• How to programmatically read an audio RTP stream using javacv and ffmpeg ?

  21 mai 2019, par Chris

  I am trying to read an audio RTP stream coming from ffmpeg in command line using javaCV. I create a DatagramSocket that listens to a specified port but can’t get the audio frames.

  I have tried with different types of buffer to play the audio to my speakers but I am getting a lot of "Invalid return value 0 for stream protocol" error messages with no audio in my speakers.

  I am running the following command to stream an audio file :

  ffmpeg -re -i /some/file.wav -ar 44100 -f mulaw -f rtp rtp ://127.0.0.1:7780

  And an excerpt of my code so far :

  public class FrameGrabber implements Runnable

  private static final TimeUnit SECONDS = TimeUnit.SECONDS;
  
  private InetAddress ipAddress;
  
  private DatagramSocket serverSocket;
  
  
  
  public FrameGrabber(Integer port) throws UnknownHostException, SocketException {
  
      super();
  
  
  
      this.ipAddress = InetAddress.getByName("192.168.44.18");
  
      serverSocket = new DatagramSocket(port, ipAddress);
  
  
  
  }
  
  
  
  public AudioFormat getAudioFormat() {
  
      float sampleRate = 44100.0F;
  
      // 8000,11025,16000,22050,44100
  
      int sampleSizeInBits = 16;
  
      // 8,16
  
      int channels = 1;
  
      // 1,2
  
      boolean signed = true;
  
      // true,false
  
      boolean bigEndian = false;
  
      // true,false
  
      return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
  
  }
  
  
  
  @Override
  
  public void run() {
  
  
  
  
  
      byte[] buffer = new byte[2048];
  
      DatagramPacket packet = new DatagramPacket(buffer, buffer.length);
  
  
  
      DataInputStream dis = new DataInputStream(new ByteArrayInputStream(packet.getData(), packet.getOffset(), packet.getLength()));
  
  
  
  
  
      FFmpegFrameGrabber grabber = new FFmpegFrameGrabber(dis);
  
      grabber.setFormat("mulaw");
  
      grabber.setSampleRate((int) getAudioFormat().getSampleRate());
  
      grabber.setAudioChannels(getAudioFormat().getChannels());
  
  
  
      SourceDataLine soundLine = null;
  
  
  
  
  
      try {
  
          grabber.start();
  
  
  
  
  
          if (grabber.getSampleRate() > 0 && grabber.getAudioChannels() > 0) {
  
  
  
              AudioFormat audioFormat = new AudioFormat(grabber.getSampleRate(), 16, grabber.getAudioChannels(), true, true);
  
  
  
              DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
  
              soundLine = (SourceDataLine) AudioSystem.getLine(info);
  
              soundLine.open(audioFormat);
  
  
  
              soundLine.start();
  
          }
  
  
  
          ExecutorService executor = Executors.newSingleThreadExecutor();
  
  
  
  
  
          while (true) {
  
  
  
              try {
  
                  serverSocket.receive(packet);
  
              } catch (IOException e) {
  
                  e.printStackTrace();
  
              }
  
  
  
              Frame frame = grabber.grab();
  
  
  
              //if (frame == null) break;
  
  
  
  
  
              if (frame != null && frame.samples != null) {
  
  
  
                  ShortBuffer channelSamplesFloatBuffer = (ShortBuffer) frame.samples[0];
  
                  channelSamplesFloatBuffer.rewind();
  
  
  
                  ByteBuffer outBuffer = ByteBuffer.allocate(channelSamplesFloatBuffer.capacity() * 2);
  
                  float[] samples = new float[channelSamplesFloatBuffer.capacity()];
  
  
  
                  for (int i = 0; i < channelSamplesFloatBuffer.capacity(); i++) {
  
                      short val = channelSamplesFloatBuffer.get(i);
  
                      outBuffer.putShort(val);
  
                  }
  
  
  
                  if (soundLine == null) return;
  
                  try {
  
                      SourceDataLine finalSoundLine = soundLine;
  
                      executor.submit(() -> {
  
                          finalSoundLine.write(outBuffer.array(), 0, outBuffer.capacity());
  
                          outBuffer.clear();
  
                      }).get();
  
                  } catch (InterruptedException interruptedException) {
  
                      Thread.currentThread().interrupt();
  
                  }
  
              }
  
  
  
          }
  
  
  
          /*
  
          executor.shutdownNow();
  
          executor.awaitTermination(1, SECONDS);
  
  
  
          if (soundLine != null) {
  
              soundLine.stop();
  
          }
  
  
  
          grabber.stop();
  
          grabber.release();*/
  
  
  
          } catch (ExecutionException ex) {
  
          System.out.println("ExecutionException");
  
          ex.printStackTrace();
  
      } catch (org.bytedeco.javacv.FrameGrabber.Exception ex) {
  
          System.out.println("FrameGrabberException");
  
          ex.printStackTrace();
  
      } catch (LineUnavailableException ex) {
  
          System.out.println("LineUnavailableException");
  
          ex.printStackTrace();
  
      }/* catch (InterruptedException e) {
  
          System.out.println("InterruptedException");
  
          e.printStackTrace();
  
      }*/
  
  
  
  
  
  }
  
  
  
  public static void main(String[] args) throws SocketException, UnknownHostException {
  
      Runnable apRunnable = new FrameGrabber(7780);
  
      Thread ap = new Thread(apRunnable);
  
      ap.start();
  
  }

  At this stage, I am trying to play the audio file in my speakers but I am getting the following logs :

  Task :FrameGrabber.main()
  Invalid return value 0 for stream protocol
  Invalid return value 0 for stream protocol
  Input #0, mulaw, from ’java.io.DataInputStream@474e6cea’ :
  Duration : N/A, bitrate : 352 kb/s
  Stream #0:0 : Audio : pcm_mulaw, 44100 Hz, 1 channels, s16, 352 kb/s
  Invalid return value 0 for stream protocol
  Invalid return value 0 for stream protocol
  Invalid return value 0 for stream protocol
  Invalid return value 0 for stream protocol
  ...

  What am I doing wrong ?

  Thanks in advance !

• java.lang.UnsatisfiedLinkError : org.bytedeco.javacpp.avutil in start FFmpegFrameGrabber

  27 février 2018, par Ashish Virani

  i have creating slowmotion video app in android and i can try to set video motion speed in any start and ending postion in total video length.

  but some error occur in start FFmpegFrameGrabber.

  here my code :

  private class PrepareMedia extends AsyncTask {
  
      private PrepareMedia() {
  
      }
  
  
  
      protected Void doInBackground(Void... params) {
  
          FFmpegFrameGrabber grabber = new FFmpegFrameGrabber(viewSource);
  
          try {
  
              grabber.start();
  
              grabber.stop();
  
              grabber.release();
  
          } catch (Exception e) {
  
              Log.e(TAG, "doInBackground: ");
  
          }
  
          return null;
  
      }
  
  }

  view my error.

  01-12 15:59:16.374 14990-14990/com.example.slowmotiondemo E/AndroidRuntime: FATAL EXCEPTION: main
  
      Process: com.example.slowmotiondemo, PID: 14990
  
      java.lang.UnsatisfiedLinkError: org.bytedeco.javacpp.avutil
  
          at java.lang.Class.classForName(Native Method)
  
          at java.lang.Class.forName(Class.java:324)
  
          at org.bytedeco.javacpp.Loader.load(Loader.java:585)
  
          at org.bytedeco.javacpp.Loader.load(Loader.java:530)
  
          at org.bytedeco.javacpp.avformat$AVFormatContext.<clinit>(avformat.java:2819)
  
          at org.bytedeco.javacv.FFmpegFrameGrabber.startUnsafe(FFmpegFrameGrabber.java:468)
  
          at org.bytedeco.javacv.FFmpegFrameGrabber.start(FFmpegFrameGrabber.java:462)
  
          at com.example.slowmotiondemo.ActivityCutMergeVideo.prepareMediaForOreantation(ActivityCutMergeVideo.java:687)
  
          at com.example.slowmotiondemo.ActivityCutMergeVideo.onCreate(ActivityCutMergeVideo.java:507)
  
          at android.app.Activity.performCreate(Activity.java:6904)
  
          at android.app.Instrumentation.callActivityOnCreate(Instrumentation.java:1136)
  
          at android.app.ActivityThread.performLaunchActivity(ActivityThread.java:3266)
  
          at android.app.ActivityThread.handleLaunchActivity(ActivityThread.java:3415)
  
          at android.app.ActivityThread.access$1100(ActivityThread.java:229)
  
          at android.app.ActivityThread$H.handleMessage(ActivityThread.java:1821)
  
          at android.os.Handler.dispatchMessage(Handler.java:102)
  
          at android.os.Looper.loop(Looper.java:148)
  
          at android.app.ActivityThread.main(ActivityThread.java:7331)
  
          at java.lang.reflect.Method.invoke(Native Method)
  
          at com.android.internal.os.ZygoteInit$MethodAndArgsCaller.run(ZygoteInit.java:1230)
  
          at com.android.internal.os.ZygoteInit.main(ZygoteInit.java:1120)
  
  </clinit>

  please tell me how to solve error.

• Script doesnt recognize bars / length right for cutting audio , ffmpeg terminal

  14 avril 2024, par totzillarbeats

  This terminal script doesn't recognize bars / length right for cutting audio, maybe somebody knows what's wrong with the calculation ...

  


  Would be happy about any help the cutting already works !

  


  #!/bin/bash

# Function to extract BPM from filename

get_bpm() {
    local filename="$1"
    local bpm=$(echo "$filename" | grep -oE '[0-9]{1,3}' | head -n1)
    echo "$bpm"
}

# Function to cut audio based on BPM
cut_audio() {
    local input_file="$1"
    local bpm="$2"
    local output_file="${input_file%.*}_cut.${input_file##*.}" # Appends "_cut" to original filename

    # Define the number of beats per bar (assuming 4 beats per bar)
    beats_per_bar=4

    # Calculate the duration of each bar in seconds
    bar_duration=$((60 * beats_per_bar / bpm))

    # Define start and end times for each bar range
    start_times=(0 21 33 45 57 69 81 93 105 117 129 141)
    end_times=(20 29 41 53 65 77 89 101 113 125 137 149)

    # Iterate through each bar range
    for ((i = 0; i < ${#start_times[@]}; i++)); do
        start_time=${start_times[$i]}
        end_time=${end_times[$i]}
        echo "Cutting audio file $input_file at $bpm BPM for bar $((i + 1)) ($start_time-$end_time) for $bar_duration seconds..."

        # Cut audio for current bar range using ffmpeg
        ffmpeg -i "$input_file" -ss "$start_time" -to "$end_time" -c copy "$output_file"_"$((i + 1)).${input_file##*.}" -y
    done

    # Check if the output files are empty and delete them if so
    for output_file in "${output_file}"_*; do
        if [ ! -s "$output_file" ]; then
            echo "Output file $output_file is empty. Deleting..."
            rm "$output_file"
        fi
    done

    echo "Audio cut and saved as $output_file"
}


# Main script
if [ "$#" -eq 0 ]; then
    echo "Usage: $0 [audio_file1] [audio_file2] ..."
    exit 1
fi

for file in "$@"; do
    bpm=$(get_bpm "$file")
    if [ -z "$bpm" ]; then
        echo "Error: No BPM found in filename $file"
    else
        cut_audio "$file" "$bpm"
    fi
done


  


  Maybe its only the math calc in the beginning but idk :)

  


  If you need more details just lmk