Custom alaw encoder does not handle media time

Question

Hi All,
I'm trying to create a custom alaw rtp packetizer but not succeeding.
I read all the rtp/alaw related forum topics and found a great help in implementing it but the time of the media is not handled with the current code.
I created custom multiplexer, packetizer, encoder based on JMF source, but the time of the media decreases 1 by every packet, and not increases 160 as it is with good 20 ms ulaw packets created by AvTransmit2.
Can you plese help me?
Thanks
Attila

Answer 1

OK, I found a solution: I wrote my own MyAudioFormat class extending the original and put duration calculation for "alaw/rtp" type of stream into computeDuration method. Then I return this new MyAudioFormat class in the getMatchingOutputFormats method of the CustomALAWPacketizer packetizer. This solved the problem and took 3,5 days to figure out.
Attila
* Implements an ALaw Packetizer.
public class CustomALAWPacketizer extends com.ibm.media.codec.audio.AudioPacketizer {
    // Methods
    public CustomALAWPacketizer() {
        packetSize=160;
            supportedInputFormats = new AudioFormat[] {
             new AudioFormat(
                    AudioFormat.ALAW,
                    8000.0,
                    8,
                    1,
                    Format.NOT_SPECIFIED,
                    Format.NOT_SPECIFIED,
                    8,
                    Format.NOT_SPECIFIED,
                    Format.byteArray
        defaultOutputFormats  = new AudioFormat[] {
            /------------------ The importand part is here -------------/
                  new MyAudioFormat (
                    MyAudioFormat.ALAW_RTP,
                    8000.0,
                    8,
                    1,
                    Format.NOT_SPECIFIED,
                    Format.NOT_SPECIFIED,
                    8,
                    Format.NOT_SPECIFIED,
                    Format.byteArray
        PLUGIN_NAME="A-Law Packetizer";
    protected  Format[] getMatchingOutputFormats(Format in) {
         return defaultOutputFormats;
    public void open() throws ResourceUnavailableException{
        setPacketSize(packetSize);
        reset();
* @(#)MyAdioFormat.java     1.40 02/08/21
* Based on Sun AudioFormat.
* Contains correct ALAW_RTP duration computation.
import javax.media.format.AudioFormat;
import javax.media.Format;
public class MyAudioFormat extends AudioFormat {
     static final long serialVersionUID = 0x12345678;
     public static String ALAW_RTP = "alaw/rtp";
     * For computing the duration of the sample.
    double multiplier = -1f;
    int margin = 0;
    boolean init = false;
    public MyAudioFormat(String encoding, double sampleRate,
                 int sampleSizeInBits, int channels,
                 int endian, int signed,
                 int frameSizeInBits, double frameRate, Class dataType) {
          super (encoding, sampleRate, sampleSizeInBits, channels, endian, signed, frameSizeInBits,
                    frameRate, dataType);
     * Returns the duration of the media based on the given length
     * of the data.
     * @param length length of the data in this format.
     * @return the duration in nanoseconds computed from the length of
     * the data in this format.  Returns -1 if the duration cannot be
     * computed.
    public long computeDuration(long length) {
     if (init) {
         // We don't know how to compute this format
         if (multiplier < 0)
          return -1;
         return (long)((length - margin) * multiplier) * 1000;
     if (encoding == null) {
         init = true;
         return -1;
     } else if (encoding.equalsIgnoreCase(AudioFormat.LINEAR) ||
             encoding.equalsIgnoreCase(AudioFormat.ULAW)) {
         if (sampleSizeInBits > 0 && channels > 0 && sampleRate > 0)
          multiplier = (1000000 * 8)/sampleSizeInBits/channels/sampleRate;
     } else if (encoding.equalsIgnoreCase(AudioFormat.ULAW_RTP)) {
         if (sampleSizeInBits > 0 && channels > 0 && sampleRate > 0)
          multiplier = (1000000 * 8)/sampleSizeInBits/channels/sampleRate;
         } else if (encoding.equalsIgnoreCase(MyAudioFormat.ALAW_RTP)) {
            /------------------ The importand part is here -------------/
         if (sampleSizeInBits > 0 && channels > 0 && sampleRate > 0)
          multiplier = (1000000 * 8)/sampleSizeInBits/channels/sampleRate;
     } else if (encoding.equalsIgnoreCase(AudioFormat.DVI_RTP)) {
         if (sampleSizeInBits > 0 && sampleRate > 0)
          multiplier = (1000000 * 8)/sampleSizeInBits/sampleRate;
         margin = 4;
        } else if (encoding.equalsIgnoreCase(AudioFormat.GSM_RTP)) {
         if (sampleRate > 0)
          multiplier = (160 * 1000000 / 33) / sampleRate;
        } else if (encoding.equalsIgnoreCase(AudioFormat.G723_RTP)) {
         if (sampleRate > 0)
          multiplier = (240/24 * 1000000) / sampleRate;
     } else if (frameSizeInBits != Format.NOT_SPECIFIED &&
             frameRate != Format.NOT_SPECIFIED) {
         // We don't know this codec, but we can compute the
         // rate by using the frame rate and size.
         if (frameSizeInBits > 0 && frameRate > 0)
          multiplier = (1000000 * 8)/frameSizeInBits/frameRate;
     init = true;
     if (multiplier > 0)
         return (long)((length - margin) * multiplier) * 1000;
     else
         return -1;
}