Screeching white sound coming while playing audio as a raw stream

Question

I. Background

1. I am trying to make an application which helps to match subtitles to the audio waveform very accurately at the waveform level, at the word level or even at the character level.
2. The audio is expected to be Sanskrit chants (Yoga, rituals etc.) which are extremely long compound words [ example - aṅganyā-sokta-mātaro-bījam is traditionally one word broken only to assist reading ]
3. The input transcripts / subtitles might be roughly in sync at the sentence/verse level but surely would not be in sync at the word level.
4. The application should be able to figure out points of silence in the audio waveform, so that it can guess the start and end points of each word (or even letter/consonant/vowel in a word), such that the audio-chanting and visual-subtitle at the word level (or even at letter/consonant/vowel level) perfectly match, and the corresponding UI just highlights or animates the exact word (or even letter) in the subtitle line which is being chanted at that moment, and also show that word (or even the letter/consonant/vowel) in bigger font. This app's purpose is to assist learning Sanskrit chanting.
5. It is not expected to be a 100% automated process, nor 100% manual but a mix where the application should assist the human as much as possible.

II. Following is the first code I wrote for this purpose, wherein

1. First I open a mp3 (or any audio format) file,
2. Seek to some arbitrary point in the timeline of the audio file // as of now playing from zero offset
3. Get the audio data in raw format for 2 purposes - (1) playing it and (2) drawing the waveform.
4. Playing the raw audio data using standard java audio libraries

III. The problem I am facing is, between every cycle there is screeching sound.

• Probably I need to close the line between cycles ? Sounds simple, I can try.
• But I am also wondering if this overall approach itself is correct? Any tip, guide, suggestion, link would be really helpful.
• Also I just hard coded the sample-rate etc ( 44100Hz etc. ), are these good to set as default presets or it should depend on the input format ?

IV. Here is the code

import com.github.kokorin.jaffree.StreamType;
import com.github.kokorin.jaffree.ffmpeg.FFmpeg;
import com.github.kokorin.jaffree.ffmpeg.FFmpegProgress;
import com.github.kokorin.jaffree.ffmpeg.FFmpegResult;
import com.github.kokorin.jaffree.ffmpeg.NullOutput;
import com.github.kokorin.jaffree.ffmpeg.PipeOutput;
import com.github.kokorin.jaffree.ffmpeg.ProgressListener;
import com.github.kokorin.jaffree.ffprobe.Stream;
import com.github.kokorin.jaffree.ffmpeg.UrlInput;
import com.github.kokorin.jaffree.ffprobe.FFprobe;
import com.github.kokorin.jaffree.ffprobe.FFprobeResult;
import java.io.IOException;
import java.io.OutputStream;
import java.nio.file.Path;
import java.nio.file.Paths;
import java.util.concurrent.TimeUnit;
import java.util.concurrent.atomic.AtomicLong;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.SourceDataLine;


public class FFMpegToRaw {
    Path BIN = Paths.get("f:\\utilities\\ffmpeg-20190413-0ad0533-win64-static\\bin");
    String VIDEO_MP4 = "f:\\org\\TEMPLE\\DeviMahatmyamRecitationAudio\\03_01_Devi Kavacham.mp3";
    FFprobe ffprobe;
    FFmpeg ffmpeg;

    public void basicCheck() throws Exception {
        if (BIN != null) {
            ffprobe = FFprobe.atPath(BIN);
        } else {
            ffprobe = FFprobe.atPath();
        }
        FFprobeResult result = ffprobe
                .setShowStreams(true)
                .setInput(VIDEO_MP4)
                .execute();

        for (Stream stream : result.getStreams()) {
            System.out.println("Stream " + stream.getIndex()
                    + " type " + stream.getCodecType()
                    + " duration " + stream.getDuration(TimeUnit.SECONDS));
        }    
        if (BIN != null) {
            ffmpeg = FFmpeg.atPath(BIN);
        } else {
            ffmpeg = FFmpeg.atPath();
        }

        //Sometimes ffprobe can't show exact duration, use ffmpeg trancoding to NULL output to get it
        final AtomicLong durationMillis = new AtomicLong();
        FFmpegResult fFmpegResult = ffmpeg
                .addInput(
                        UrlInput.fromUrl(VIDEO_MP4)
                )
                .addOutput(new NullOutput())
                .setProgressListener(new ProgressListener() {
                    @Override
                    public void onProgress(FFmpegProgress progress) {
                        durationMillis.set(progress.getTimeMillis());
                    }
                })
                .execute();
        System.out.println("audio size - "+fFmpegResult.getAudioSize());
        System.out.println("Exact duration: " + durationMillis.get() + " milliseconds");
    }

    public void toRawAndPlay() throws Exception {
        ProgressListener listener = new ProgressListener() {
            @Override
            public void onProgress(FFmpegProgress progress) {
                System.out.println(progress.getFrame());
            }
        };

        // code derived from : https://stackoverflow.com/questions/32873596/play-raw-pcm-audio-received-in-udp-packets

        int sampleRate = 44100;//24000;//Hz
        int sampleSize = 16;//Bits
        int channels   = 1;
        boolean signed = true;
        boolean bigEnd = false;
        String format  = "s16be"; //"f32le"

        //https://trac.ffmpeg.org/wiki/audio types
        final AudioFormat af = new AudioFormat(sampleRate, sampleSize, channels, signed, bigEnd);
        final DataLine.Info info = new DataLine.Info(SourceDataLine.class, af);
        final SourceDataLine line = (SourceDataLine) AudioSystem.getLine(info);

        line.open(af, 4096); // format , buffer size
        line.start();

        OutputStream destination = new OutputStream() {
            @Override public void write(int b) throws IOException {
                throw new UnsupportedOperationException("Nobody uses thi.");
            }
            @Override public void write(byte[] b, int off, int len) throws IOException {
                String o = new String(b);
                boolean showString = false;
                System.out.println("New output ("+ len
                        + ", off="+off + ") -> "+(showString?o:"")); 
                // output wave form repeatedly

                if(len%2!=0) {
                    len -= 1;
                    System.out.println("");
                }
                line.write(b, off, len);
                System.out.println("done round");
            }
        };

        // src : http://blog.wudilabs.org/entry/c3d357ed/?lang=en-US
        FFmpegResult result = FFmpeg.atPath(BIN).
            addInput(UrlInput.fromPath(Paths.get(VIDEO_MP4))).
            addOutput(PipeOutput.pumpTo(destination).
                disableStream(StreamType.VIDEO). //.addArgument("-vn")
                setFrameRate(sampleRate).            //.addArguments("-ar", sampleRate)
                addArguments("-ac", "1").
                setFormat(format)              //.addArguments("-f", format)
            ).
            setProgressListener(listener).
            execute();

        // shut down audio
        line.drain();
        line.stop();
        line.close();

        System.out.println("result = "+result.toString());
    }

    public static void main(String[] args) throws Exception {
        FFMpegToRaw raw = new FFMpegToRaw();
        raw.basicCheck();
        raw.toRawAndPlay();
    }
}

Thank You

Answer 1

I suspect your screech sound stems from a half-filled buffer that is handed to the audio system.

As indicated in the comment above, I'd use something like FFSampledSP (if on mac or Windows) and then code like the following, which is much more java-esque.

Just make sure the FFSampledSP complete jar is in your path and you should be good to go.

import javax.sound.sampled.*;
import java.io.File;
import java.io.IOException;

public class PlayerDemo {

    /**
     * Derive a PCM format.
     */
    private static AudioFormat toSignedPCM(final AudioFormat format) {
        final int sampleSizeInBits = format.getSampleSizeInBits() <= 0 ? 16 : format.getSampleSizeInBits();
        final int channels = format.getChannels() <= 0 ? 2 : format.getChannels();
        final float sampleRate = format.getSampleRate() <= 0 ? 44100f : format.getSampleRate();
        return new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
                sampleRate,
                sampleSizeInBits,
                channels,
                (sampleSizeInBits > 0 && channels > 0) ? (sampleSizeInBits/8)*channels : AudioSystem.NOT_SPECIFIED,
                sampleRate,
                format.isBigEndian()
        );
    }


    public static void main(final String[] args) throws IOException, UnsupportedAudioFileException, LineUnavailableException {
        final File audioFile = new File(args[0]);
        // open mp3 or whatever
        final Long durationInMicroseconds = (Long)AudioSystem.getAudioFileFormat(audioFile).getProperty("duration");
        // how long is the file, use AudioFileFormat properties
        System.out.println("Duration in microseconds (not millis!): " + durationInMicroseconds);
        // open the mp3 stream (not yet decoded)
        final AudioInputStream mp3In = AudioSystem.getAudioInputStream(audioFile);
        // derive a suitable PCM format that can be played by the AudioSystem
        final AudioFormat desiredFormat = toSignedPCM(mp3In.getFormat());
        // ask the AudioSystem for a source line for playback
        // that corresponds to the derived PCM format
        final SourceDataLine line = AudioSystem.getSourceDataLine(desiredFormat);

        // now play, typically in separate thread
        new Thread(() -> {
            final byte[] buf = new byte[4096];
            int justRead;
            // convert to raw PCM samples with the AudioSystem
            try (final AudioInputStream rawIn = AudioSystem.getAudioInputStream(desiredFormat, mp3In)) {
                line.open();
                line.start();
                while ((justRead = rawIn.read(buf)) >= 0) {
                    // only write bytes we really read, not more!
                    line.write(buf, 0, justRead);
                    final long microsecondPosition = line.getMicrosecondPosition();
                    System.out.println("Current position in microseconds: " + microsecondPosition);
                }
            } catch (IOException | LineUnavailableException e) {
                e.printStackTrace();
            } finally {
                line.drain();
                line.stop();
            }
        }).start();
    }
}

The regular Java API does not allow to jump to arbitrary positions. However, FFSampledSP contains an extension, i.e. a seek() method. To use it, just cast the rawIn from the example above to FFAudioInputStream and call seek() with a time and a timeUnit.