Eu estou tentando implementar um sistema onde eu dou um conjunto de freqüências para ser jogado de uma vez, atualmente posso jogar cada freqüência individualmente. Abaixo eu tenho um código, que reproduz a frequência dada, um de cada vez.

    import java.applet.*;
    import java.io.*;
    import java.net.*;
    import javax.sound.sampled.*;

    public final class StdAudio {

        public static final int SAMPLE_RATE = 44100;

        private static final int BYTES_PER_SAMPLE = 2;                // 16-bit audio
        private static final int BITS_PER_SAMPLE = 16;                // 16-bit audio
        private static final double MAX_16_BIT = Short.MAX_VALUE;     // 32,767
        private static final int SAMPLE_BUFFER_SIZE = 4096;


        private static SourceDataLine line;   // to play the sound
        private static byte[] buffer;         // our internal buffer
        private static int bufferSize = 0;    

        // not-instantiable
        private StdAudio() { }


        // static initializer
        static { init(); }

        // open up an audio stream
    private static void init() {
    try {
        // 44,100 samples per second, 16-bit audio, mono, signed PCM, little Endian
        AudioFormat format = new AudioFormat((float) SAMPLE_RATE, BITS_PER_SAMPLE, 1, true, false);
        DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);

        line = (SourceDataLine) AudioSystem.getLine(info);
        line.open(format, SAMPLE_BUFFER_SIZE * BYTES_PER_SAMPLE);

        buffer = new byte[SAMPLE_BUFFER_SIZE * BYTES_PER_SAMPLE/3];
    } catch (Exception e) {
        System.out.println(e.getMessage());
        System.exit(1);
    }

    // no sound gets made before this call
    line.start();
}


/**
 * Close standard audio.
 */
public static void close() {
    line.drain();
    line.stop();
}

/**
 * Write one sample (between -1.0 and +1.0) to standard audio. If the sample
 * is outside the range, it will be clipped.
 */
public static void play(double in) {

    // clip if outside [-1, +1]
    if (in < -1.0) in = -1.0;
    if (in > +1.0) in = +1.0;

    // convert to bytes
    short s = (short) (MAX_16_BIT * in);
    buffer[bufferSize++] = (byte) s;
    buffer[bufferSize++] = (byte) (s >> 8);   // little Endian

    // send to sound card if buffer is full        
    if (bufferSize >= buffer.length) {
        line.write(buffer, 0, buffer.length);
        bufferSize = 0;
    }
}

/**
 * Write an array of samples (between -1.0 and +1.0) to standard audio. If a sample
 * is outside the range, it will be clipped.
 */
public static void play(double[] input) {
    for (int i = 0; i < input.length; i++) {
        play(input[i]);
    }
}

private static double[] tone(double hz, double duration) {
    int N = (int) (StdAudio.SAMPLE_RATE * duration);
    double[] a = new double[N+1];
    for (int i = 0; i <= N; i++)
        a[i] = amplitude * Math.sin(2 * Math.PI * i * hz / StdAudio.SAMPLE_RATE);
    return a;
}

/**
 * Test client - play an A major scale to standard audio.
 */
public static void main(String[] args) {


    double hz = 440.4// 440.4 Hz for 1 sec
    double duration = 1.0;

    double[] a = tone(hz,duration);
    StdAudio.play(a);

}

}