Setting RTP Stream Properties FrameRate/Size/Format

i am having problems setting the framerate on the rtp stream, i can set the quality of the jpeg, set the size of the stream, ive tried sizes 160X120, 320X240, 640x480.
as soon as i try to set the framerate it dies.
any ideas?

Similar Messages

Setting in itunes for sync file size/format

I heard there was a setting in itunes the would allow you to pick a smaller files size/format at the time of syncing to ypour ipod so during the sync the files would be compressed. Is this true. My cd collection is ripped and saved in lossless and i was thinking of syncing at 320kbps to see if i noticed a difference. It would be nice if this was ture so i can save all my cds at lossless and just compress when i sync if i want.
Thanks!!!

So that will compress all the larger files on the ipod to 128kbps ACC....then i can load more lossless till memory is full again and, do that again, and it will compress those new larger files to 128kbps AAC????
Also, can you pick another file size. I already know i loath the sound quality of 128. I was hoping to do something like this but choose 320

Write(Export) the RTP Stream from a SIP Call to an audio file

I am working on telephony system that need record each call to a file.
I got the RTP Stream in ReceiveStreamEvent handler and start to record
the problem is i got a file and play it but there are any sounds
when i set FileTypeDescriptor.WAVE and AudioFormat.ULAW
I try out the FileTypeDescriptor.WAVE and AudioFormat.IMA4_MS, i got a fixed size file with 60kb
if the FileTypeDescriptor.MPEG_AUDIO and AudioFormat.MPEG, the processor cannot be realize,
the thread is blocked!!
Could anyone save me? thanks in advanced!
=================================Code===================================================
public void update(ReceiveStreamEvent evt){
logger.debug("received a new incoming stream. " + evt);
RTPManager mgr = (RTPManager) evt.getSource();
Participant participant = evt.getParticipant(); // could be null.
ReceiveStream stream = evt.getReceiveStream(); // could be null.
if (evt instanceof NewReceiveStreamEvent)
try
stream = ( (NewReceiveStreamEvent) evt).getReceiveStream();
DataSource ds = stream.getDataSource();
          recorder = new CallRecorderImpl(ds);
recorder.start();
catch (Exception e)
logger.error("NewReceiveStreamEvent exception ", e);
return;
========================================================================================
this is the CallRecorderImpl Class
public class CallRecorderImpl implements DataSinkListener, ControllerListener {
     private Processor processor = null;
     private DataSink dataSink = null;
     private DataSource source = null;
     private boolean bRecording = false;
     FileTypeDescriptor contentType = new FileTypeDescriptor(FileTypeDescriptor.WAVE);
     Format encoding = new AudioFormat(AudioFormat.IMA4_MS);
MediaLocator dest = new MediaLocator("file:/c:/bar.wav");
     public CallRecorderImpl(DataSource ds) {
          this.source = ds;
     public void start() {
          try {
               processor = Manager.createProcessor(source);
               processor.addControllerListener(this);
               processor.configure();
          } catch (Exception e) {
               System.out.println("exception:" + e);
     public void stop() {
          try {
               System.out.println("stopping");
               this.dataSink.stop();
               this.dataSink.close();
          } catch (Exception ep) {
               ep.printStackTrace();
     public void controllerUpdate(ControllerEvent evt) {
          Processor pr = (Processor) evt.getSourceController();
          if (evt instanceof ConfigureCompleteEvent) {
               System.out.println("ConfigureCompleteEvent");
               processConfigured(pr);
          if (evt instanceof RealizeCompleteEvent) {
               System.out.println("RealizeCompleteEvent");
               processRealized(pr);
          if (evt instanceof ControllerClosedEvent) {
               System.out.println("ControllerClosedEvent");
          if (evt instanceof EndOfMediaEvent) {
               System.out.println("EndOfMediaEvent");
               pr.stop();
          if (evt instanceof StopEvent) {
               System.out.println("StopEvent");
               pr.close();
               try {
                    dataSink.stop();
                    dataSink.close();
               } catch (Exception ee) {
                    ee.printStackTrace();
     public void dataSinkUpdate(DataSinkEvent event) {
          if (event instanceof EndOfStreamEvent) {
               try {
                    System.out.println("EndOfStreamEvent");
                    dataSink.stop();
                    dataSink.close();
                    System.exit(1);
               } catch (Exception e) {
     public void processConfigured(Processor p) {
          // Set the output content type
          p.setContentDescriptor(this.contentType);
          // Get the track control objects
          TrackControl track[] = p.getTrackControls();
          boolean encodingPossible = false;
          // Go through the tracks and try to program one of them
          // to output ima4 data.
          for (int i = 0; i < track.length; i++) {
          try {
               track.setFormat(this.encoding);
               encodingPossible = true;
          } catch (Exception e) {
               track[i].setEnabled(false);
          if (!encodingPossible) {
               System.out.println("cannot encode to " + this.encoding);
          return;
          p.realize();
     public void processRealized(Processor p) {
          System.out.println("Entered processRealized");
DataSource source = p.getDataOutput();
try {
     dataSink = Manager.createDataSink(source, dest);
     dataSink.open();
     dataSink.start();
     dataSink.addDataSinkListener(new DataSinkListener() {
                    public void dataSinkUpdate(DataSinkEvent e) {
                         if (e instanceof EndOfStreamEvent) {
                              System.out.println("EndOfStreamEvent");
                              dataSink.close();
} catch (Exception e) {
     e.printStackTrace();
     return;
          p.start();
============================================

Which shows that the output stream cannot be of that particulat format and descriptor.
Look at this code
import javax.swing.*;
import javax.media.*;
import java.net.*;
import java.io.*;
import javax.media.datasink.*;
import javax.media.protocol.*;
import javax.media.format.*;
import javax.media.control.*;
class Abc implements ControllerListener
     DataSource ds;
     DataSink sink;
     Processor pr;
     MediaLocator mc;
     public void maam() throws Exception
          mc=new MediaLocator("file:C:/Workspace/aaaaa.mpg");
          pr=Manager.createProcessor(new URL("file:G:/java files1/jmf/aa.mp3"));
          pr.addControllerListener(this);
          pr.configure();
     public void controllerUpdate(ControllerEvent e)
          if(e instanceof ConfigureCompleteEvent)
               System.out.println ("ConfigureCompleteEvent");
                         Processor p = (Processor)e.getSourceController();
               System.out.println ("ConfigureCompleteEvent");
                     processConfigured(p);
          if(e instanceof RealizeCompleteEvent)
               System.out.println ("RealizeCompleteEvent");
                       Processor p = (Processor)e.getSourceController();
                         processRealized(p);
          if(e instanceof ControllerClosedEvent)
                     System.out.println ("ControllerClosedEvent");
                 if(e instanceof EndOfMediaEvent)
                    System.out.println ("EndOfMediaEvent");
                    Processor p = (Processor)e.getSourceController();
                    p.stop();
             if(e instanceof StopEvent)
                     System.out.println ("StopEvent");
                     Processor p = (Processor)e.getSourceController();
                     p.close();
               try
                    sink.stop();
                    sink.close();
               catch(Exception ee)
     public void processConfigured(Processor p)
          System.out.println("Entered processConfigured");
          p.setContentDescriptor (new FileTypeDescriptor (FileTypeDescriptor.MPEG_AUDIO));
            /*    TrackControl track[] = p.getTrackControls ();
             boolean encodingPossible = false;
             for (int i = 0; i < track.length; i++)
                  try
                       track.setFormat (new VideoFormat (VideoFormat.MPEG));
               encodingPossible = true;
               catch (Exception e)
               track[i].setEnabled (false);
     p.realize();
     public void processRealized(Processor p)
          pr=p;
          System.out.println("Entered processRealized");
          try
               MediaLocator dest = new MediaLocator("file:C:/Workspace/ring1.mpg");
               sink = Manager.createDataSink(p.getDataOutput(), dest);
          sink.addDataSinkListener(new DataSinkListener()
                    public void dataSinkUpdate(DataSinkEvent e)
                         if(e instanceof EndOfStreamEvent)
                    System.out.println ("EndOfStreamEvent");
                              sink.close();
               sink.open();
               sink.start();
          catch(Exception eX)
          System.out.println("Just before start");
          p.start();
/*     public void dataSinkUpdate(DataSinkEvent event)
          if(event instanceof EndOfStreamEvent)
               try
                    System.out.println("EndOfStreamEvent");
                    dsk.stop();
                    dsk.close();
                    System.exit(1);
               catch(Exception e)
public class JMFCapture6
     public static void main(String args[]) throws Exception
          Abc a=new Abc();
          a.maam();

Merging RTP streams and writing to file

I'm am in the middle of project using JMF i'm trying to merge incoming RTP streams and write them to file but i always have problem with formats. I have created processor with merged DataSource i set ContentDescriptor to different FileTypeDescriptors and tried to transcode the RTP streams to different formats, but always processor couldnt realize because of this error:
Unable to handle format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Unable to handle format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Could someone tell me what content descriptors i need to use and what formats to solve this problem. Any help will be appreciated.
Message was edited by:
kaligula

Hello,
Here is a procedure i wrote to write on text files the content of any table (and also generate the insert SQL order)
http://oracle.developpez.com/sources/?page=developpement#Extraction_table
You can download the script file here http://sheikyerbouti.developpez.com/src/extraction_table.zip
Hope this help you.
Francois

Attempting to write RTP stream to wave

I have been working long and hard trying to figure out to save an RTP stream from a Cisco phone to a wave. I need the wave file to be in the format ULAW, 8khz, 8 bit mono.
Through much research this is the code I come up with but I does not work. I states that I am trying to get a datasink for null when I try to create the filewriter datasink.
I also have gotten the following error message with other version of the code below:
newReceiveStreamEvent exception Cannot find a DataSink for: com.sun.media.protocol.rtp.DataSource@3680c1
When I get the RTP stream I get the following as the format of the input stream:
- Recevied new RTP stream: ULAW/rtp, 8000.0 Hz, 8-bit, Mono
else if (evt instanceof NewReceiveStreamEvent) {
     try {
          stream = ((NewReceiveStreamEvent)evt).getReceiveStream();
Format formats[] = new Format[1];
formats[0] = new AudioFormat(AudioFormat.LINEAR,8000,8,1);
FileTypeDescriptor outputType = new FileTypeDescriptor(FileTypeDescriptor.WAVE);
DataSource ds = stream.getDataSource();
          // Find out the formats.
          RTPControl ctl = (RTPControl)ds.getControl("javax.media.rtp.RTPControl");
          if (ctl != null){
          System.err.println(" - Recevied new RTP stream: " + ctl.getFormat());
          } else
          System.err.println(" - Recevied new RTP stream");
          if (participant == null)
          System.err.println(" The sender of this stream had yet to be identified.");
          else {
          System.err.println(" The stream comes from: " + participant.getCNAME());
ProcessorModel processorModel = new ProcessorModel(ds, formats, null);
Processor processor = Manager.createRealizedProcessor(processorModel);
          processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW_RTP));
MediaLocator f = new MediaLocator("file:/C:/output.wav");
DataSink filewriter = null;
DataSource tempds = processor.getDataOutput();
filewriter = Manager.createDataSink(tempds, f);
filewriter.open();
Edited by: phodat on Sep 16, 2010 10:17 PM
Edited by: phodat on Sep 16, 2010 10:20 PM

The content descriptor is designed to specify the output format, not the input format. The input format is specified by the DataSource automatically, and you set the content descriptor to whatever you want the output format to be. In this case, you'd want to set it to a WAV file.
You'd then need to go through all of the Track objects on the processor and set their output format to the ULAW specifications as you want there.
[http://www.cs.odu.edu/~cs778/spring04/lectures/jmfsolutions/examplesindex.html]
There's actually an example program that exports RTP streams, you can probably use that code without modification to fit your needs.

RTP streaming noise from JMStudio

Am using the AVTransmit3 (also tried AVTransmit2) downloaded code to send a RTP stream to JMStudio. It does recognize it as ULAW as does Ethereal however JMStudio only plays noise. This is a priority project so any help would be much appreciated.
java AVTransmit3 rtp:172.16.85.2:8000/audio 192.168.10.223 9000
Track 0 is set to transmit as:
ULAW/rtp, 8000.0 Hz, 8-bit, Mono
Created RTP session: 192.168.10.223 9000
Start transmission for 60 seconds...
...transmission ended.

Change the video format to JPEG/RTP

How to extract data from Buffer and create a RTP stream

Hi
I'm working on a project where I need to interrupt a media stream (video and audio) and extract the data. Send the data with a custom protocol. On the recieveing side I would like to reconstruct the stream using only the data chunks.
I'm currently looking at [DataSourceReader.java|http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/DataSourceReader.java] and more specifically at a method like printDataInfo(Buffer buffer) to extract the data.
Is it possible to create a RTP stream, only having access to the byte array "data" in Buffer ?
Thanks in advance.

camelstrike wrote:
Hi
I'm working on a project where I need to interrupt a media stream (video and audio) and extract the data. Send the data with a custom protocol. On the recieveing side I would like to reconstruct the stream using only the data chunks.
I'm currently looking at [DataSourceReader.java|http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/DataSourceReader.java] and more specifically at a method like printDataInfo(Buffer buffer) to extract the data.
There are a couple of different ways to get the data. Reading it from inside a DataSink is perfectly fine...
Is it possible to create a RTP stream, only having access to the byte array "data" in Buffer ?Yes and no.
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/CustomPayload.html]
You need to know the format of the media in addition to the actual media data...

Set-up custom font, font size, and color for writing new and reply e-mails in Thunderbird

I know that I can select the font, size, and color for writing each new and reply e-mails in Thunderbird, but I don't want to have to do this each time. I want to set-up the default font, size, and color for all new and reply e-mails. It already comes up with a default one, but I want to change it, but can't figure out how to do this. I'm sure it is quite simple, but I don't know how to do this. Thank you in advance for help.

Tools > Options > Display > Formatting tab
Default font: select font
Select size : eg; 14
Click on 'Advanced' button and set all the sizes to 14
Select : 'allow messages to use other fonts'
click on OK
Click on 'Composition'
Under the 'General' tab
Suggest you set HTML font to 'Variable width' and Size : 'Medium'
Select the 'Text colour' you would like to compose email using
Select the 'Background colour' you would like to use when composing emails.
click on OK to save all changes.
This will set the display email list to the selected font.
It will allow received emails to use the font the sender used.
When you compose an email it will also use the same font settings.
The message 'Display' settings in the above section also apply to messages that you compose. The settings are not sent, so they do not affect how your messages appear to recipients.
The settings in the ''Composition' section can affect how messages are sent. If you make unusual choices, then people who receive messages from you might find them difficult or impossible to read. Hence, why I suggest you leave it as 'Variable width' and allow the 'Display' settings to set the default used for composing emails.
More info:
* http://kb.mozillazine.org/Font_settings_in_Thunderbird

Writing a conference server for RTP streams

Hello,
I'm trying to write a conference server which accepts multiple RTP streams (one for each participant), creates a mixed RTP stream of all other participants and sends that stream back to each participant.
For 2 participants, I was able to correctly receive and send the stream of the other participant to each party.
For 3 participants, creating the merging data source does not seem to work - i.e. no data is received by the participants.
I tried creating a cloneable data sources instead, thinking that this may be the root cause, but when creating cloneable data sources from incoming RTP sources, I am unable to get the Processor into Configured state, it seems to deadlock. Here's the code outline :
    Iterator pIt = participants.iterator();
    List dataSources = new ArrayList();
    while(pIt.hasNext()) {
      Party p = (Party) pIt.next();
      if(p!=dest) {
        DataSource ds = p.getDataSource();
        DataSource cds = Manager.createCloneableDataSource(ds);
        DataSource clone= ((SourceCloneable)cds).createClone();
        dataSources.add(clone);
    Object[] sources = dataSources.toArray(new DataSource[0]);
    DataSource dataSource =   Manager.createMergingDataSource((DataSource[])sources);
    Processor p = Manager.createProcessor(dataSource);
    MixControllerListener cl = new MixControllerListener();
    p.addControllerListener(cl);
    // Put the Processor into configured state.
    p.configure();
    if (!cl.waitForState(p, p.Configured)) {
        System.err.println("Failed to configure the processor.");
        assert false;
    }Here are couple of stack traces :
"RTPEventHandler" daemon prio=1 tid=0x081d6828 nid=0x3ea6 in Object.wait() [98246000..98247238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f37e4a8> (a java.lang.Object)
        at java.lang.Object.wait(Object.java:429)
        at demo.Mixer$MixControllerListener.waitForState(Mixer.java:248)
        - locked <0x9f37e4a8> (a java.lang.Object)
        at demo.Mixer.createMergedDataSource(Mixer.java:202)
        at demo.Mixer.createSendStreams(Mixer.java:165)
        at demo.Mixer.createSendStreamsWhenAllJoined(Mixer.java:157)
        - locked <0x9f481840> (a demo.Mixer)
        at demo.Mixer.update(Mixer.java:123)
        at com.sun.media.rtp.RTPEventHandler.processEvent(RTPEventHandler.java:62)
        at com.sun.media.rtp.RTPEventHandler.dispatchEvents(RTPEventHandler.java:96)
        at com.sun.media.rtp.RTPEventHandler.run(RTPEventHandler.java:115)
"JMF thread: com.sun.media.ProcessEngine@a3c5b6[ com.sun.media.ProcessEngine@a3c5b6 ] ( configureThread)" daemon prio=1 tid=0x082fe3c8 nid=0x3ea6 in Object.wait() [977e0000..977e1238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f387560> (a java.lang.Object)
        at java.lang.Object.wait(Object.java:429)
        at com.sun.media.parser.RawBufferParser$FrameTrack.parse(RawBufferParser.java:247)
        - locked <0x9f387560> (a java.lang.Object)
        at com.sun.media.parser.RawBufferParser.getTracks(RawBufferParser.java:112)
        at com.sun.media.BasicSourceModule.doRealize(BasicSourceModule.java:180)
        at com.sun.media.PlaybackEngine.doConfigure1(PlaybackEngine.java:229)
        at com.sun.media.ProcessEngine.doConfigure(ProcessEngine.java:43)
        at com.sun.media.ConfigureWorkThread.process(BasicController.java:1370)
        at com.sun.media.StateTransitionWorkThread.run(BasicController.java:1339)
"JMF thread" daemon prio=1 tid=0x080db410 nid=0x3ea6 in Object.wait() [97f41000..97f41238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f480578> (a com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave)
        at java.lang.Object.wait(Object.java:429)
        at com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave.run(CloneableSourceStreamAdapter.java:375)
        - locked <0x9f480578> (a com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave)
        at java.lang.Thread.run(Thread.java:534)Any ideas ?
Thanks,
Jarek

bgl,
I was able to get past the cloning issue by following the Clone.java example to the letter :)
Turns out that the cloneable data source must be added as a send stream first, and then the clonet data source. Now for each party in the call the conf. server does the following :
Party(RTPManager mgr,DataSource ds) {
      this.mgr=mgr;
      this.ds=Manager.createCloneableDataSource(ds);
   synchronized DataSource cloneDataSource() {
      DataSource retVal;
      if(getNeedsCloning()) {
        retVal = ((SourceCloneable) ds).createClone();
      } else {
        retVal = ds;
        setNeedsCloning();
      return retVal;
    private void setNeedsCloning() {
      needsCloning=true;
    private boolean getNeedsCloning() {
      return needsCloning;
     private synchronized void addSendStreamFromNewParticipant(Party newOne) throws UnsupportedFormatException, IOException {
    debug("*** - New one joined. Creating the send streams. Curr count :" + participants.size());
    Iterator pIt = participants.iterator();
    while(pIt.hasNext()) {
      Party p = (Party)pIt.next();
      assert p!=newOne;
      // update existing participant
      SendStream sendStream = p.getMgr().createSendStream(newOne.cloneDataSource(),0);
      sendStream.start();
      // send data from existing participant to the new one
      sendStream = newOne.getMgr().createSendStream(p.cloneDataSource(),0);
      sendStream.start();
    debug("*** - Done creating the streams.");So I made some progress, but I'm still not quite there.
The RTP manager JavaDoc for createSendStream states the following :
* This method is used to create a sending stream within the RTP
* session. For each time the call is made, a new sending stream
* will be created. This stream will use the SDES items as entered
* in the initialize() call for all its RTCP messages. Each stream
* is sent out with a new SSRC (Synchronisation SouRCe
* identifier), but from the same participant i.e. local
* participant. <BR>
For 3 participants, my conf. server creates 2 send streams to every one of them, so I'd expect 2 SSRCs on the wire. Examining the RTP packets in Ethereal, I only see 1 SSRC, as if the 2nd createSendStream call failed. Consequently, each participany in the conference is able to receive voice from only 1 other participant, even though I create RTPManager instance for each participany, and add 2 send streams.
Any ideas ?
Thanks,
Jarek

IPhone viewing Streaming Video in other formats

I am considering an iPhone, however, I am curious if the iPhone can view streaming video in other formats such as wmv. I have tried on a friends phone and it didn't work. Is there a plug-in for the iPhone that can be installed to do this?

My suggestion is to go to the feedback site (which I will list below) and tell Apple your suggestions directly. Not a promise that they will be included but at least you are getting to the right set of ears.
http://www.apple.com/feedback/iphone.html

Set default font type and size in formcreated using Acrobat X Pro

I have a word document that I used Acrobat X Pro to create a PDF and then a Form. The Form has over 200 text fields. Is there a way to globally set the font type and size to one type, for instance Arial 10 so that when someone opens the Form and fills it in, the global font type and size is preset

Are you talking about fields that you create manually or field created with the Form Wizard? Exactly what version of Acrobat are you using? What OS?
In any case, using a script is a quick and easy approach:
for (var i = 0; i < numFields; i++) {
    var fName = getNthFieldName(i);
    var f = getField(fName);
    if (f.type === "text") {
        f.textFont = font.Times;
        f.textSize = 9;
        // Other properties go here
I use this type of thing all the time to quickly alter a form. Such a script isn't added anywhere, just executed (e.g., JavaScript console). Over time a developer can build up a collection of utility scripts that can do all sorts of things to dramatically speed up development. It would be bad if this option were not available.
If there is a bug, it should certainly be addressed, so don't think I'm arguing against that, but I'm not seeing it on any of my systems. I can successfully set the default field properties by selecting "Use Current Properties as New Defaults". The Form Wizard does not use these defaults, but that's a different matter.

How to synchronize audio and video rtp-streams

Hi everyone!
I've tried to make one of the processors, processor1, control the other processor, processor2, with processor1.addController(processor2) but I get a
javax.media.IncompatibleTimeBaseException. I need to synchronize audio and video rtp-streams because they are sometimes completely out of sync.
In JMF API Guide I've read:
"2. Determine which Player object's time base is going to be used to drive
the other Player objects and set the time base for the synchronized
Player objects. Not all Player objects can assume a new time base.
For example, if one of the Player objects you want to synchronize has
a *push-data-source*, that Player object's time base must be used to
drive the other Player objects."
I'm using a custom AVReceive3 to receive rtp-streams and then I create processors for the incoming stream's datasource's, and they are ALL PushBufferDataSource's.
Does this mean I can't synchronize these. If so is there any way I can change them into Pull...DataSources ?
How are you supposed to sync audio and video if not with as above ?

camelstrike wrote:
Hi everyone!
I've tried to make one of the processors, processor1, control the other processor, processor2, with processor1.addController(processor2) but I get a
javax.media.IncompatibleTimeBaseException. I need to synchronize audio and video rtp-streams because they are sometimes completely out of sync.
In JMF API Guide I've read:
"2. Determine which Player object's time base is going to be used to drive
the other Player objects and set the time base for the synchronized
Player objects. Not all Player objects can assume a new time base.
For example, if one of the Player objects you want to synchronize has
a *push-data-source*, that Player object's time base must be used to
drive the other Player objects."
I'm using a custom AVReceive3 to receive rtp-streams and then I create processors for the incoming stream's datasource's, and they are ALL PushBufferDataSource's.
Does this mean I can't synchronize these. If so is there any way I can change them into Pull...DataSources ?The RTP packets are timestamped when they leave, and they are played in order. You can't change the timebase on an RTP stream because, for all intensive purposes, it's "live" data. It plays at the speed the transmitting RTP server wants it to play, and it never gets behind because it drops out of order and old data packets.
If your RTP streams aren't synced correctly on the receiving end, it means they aren't being synced on the transmitting end...so you should look into the other side of the equation.

How can I manage controls in an RTP Streaming

Hello,
I'm currently using an RTP Streaming in order to play some music. I started with AVReceive2 and AVTransmit2 from sun and by adding a plugin, I managed to play MP3. So far so good.
Because I didn't want to use view components given by sun, I created my own GUI for my player and I use it instead of using the one in sun's examples.
I use listeners for my "buttons" (I prefer using mouselistener and Panel like in the sun's example Jamp) but I can't find a way to execute those buttons.
If I press pause, it has to pause the streaming in AVTransmit2.
If I press next, or previous, it has to change the process AVTransmit2.
Each control has to be done in AVTransmit2 but my player is part of AVReceive2.
My first guess was to check if I could use an event to tell my AVTransmit2 object to execute my controls but I havn't found a way to tell my AVTransmit2 object that it was this button I pressed.
I finally stopped trying transmitting my orders thanks to the events.
I tried then to find a way to get my AVTransmit2 object in my AVReceive2 object so that I could call methods but i failed :(
How can i manage my controls so that it will call methods on my server and not my client ?
Thanks :)
Shad.
Edited by: Shadwolf on Feb 9, 2010 2:15 AM

Hi, again,
For the next & previous button, I was thinking in something.
Do you think it could be good to create a new class RTPClientManager which extends from RTPManager and get 2 booleans previous & next that are set to true when I press on buttons ?
From here, couldn't I modify my function in AVTransmit2 like this (I implements ReseiveStreamEvent on AVTransmit2) :
@Override
     public void update(ReceiveStreamEvent evt)
          RTPClientManager mgr = (RTPClientManager )evt.getSource();
          Participant participant = evt.getParticipant();     // could be null.
          ReceiveStream stream = evt.getReceiveStream(); // could be null.
           * Détection de la fermeture de connexion du client afin de fermer la transmission streaming
          if (evt instanceof ByeEvent)
               System.err.println(" - Got \"bye\" from: " + participant.getCNAME());
                     if(mgr.isNext())
                 this.stop();
                         /* SOME STUFF FOR NEXT */
                    else if(mgr.isPrevious())
                 this.stop();
                         /* SOME STUFF FOR PREVIOUS*/
                else //means it's the stop button called
                          this.stop() ;
     }Would it work ?
If you have better ideas I am ready to hear theam :P
But if this works, I will manage stop, next and previous, but how could I manage play & pause button ?
I can't find a proper solution to manage my controls :(

Stream to file size

I need a way that I can stream audio to a file and set a cap on the size but never stop putting new audio in the file. Is there a way that I can to this. I know already how to send audio to a file but not cap it and continue.
--John

Use a random access file and write data to start of the file again once the limit has been reached. When finished you would need to copy this data so the start is in the right place.

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Hi,
I am playing an h.263/rtp stream which is multicasted over a network. I want to play the file with rotation or some other effect added to it. I tried to see following link for some help(as the forums suggested):- http://java.sun.com/products/java-media/jmf/2.1.1/solutions/RotationEffect.html. But the link is disabled and new link which they are providing does not contain the example.
Can any of me help me how to add the effect on h263/rtp.
Though i already created my effect class and tried adding it. But could not add the effect.
Kindly give me some directions.

Is there some error in above code or if there is some other issue.There's an error in the above code because you're assuming I'm being imprecise with my language. I am not.
You must copy the data from the input buffer to the output buffer...
Buffer objects are wrappers around byte[], and you cannot copy data from one array to another by modifying the object references...which is what your code does.
public int process(Buffer inBuffer, Buffer outBuffer) {
    // Make sure the output buffer will hold data
    if (!(outBuffer.getData() instanceof byte[])) {
        outBuffer.setData(new byte[inBuffer.getLength()]);
        outBuffer.setLength(inBuffer.getLength());
        outBuffer.setOffset(0);
    // Make sure the output buffer is long enough
    if (outBuffer.getLength() + outBuffer.getOffset() < inBuffer.getLength())) {
        int offset = outBuffer.getOffset();
        int length = inBuffer.getLength() + outBuffer.getOffset();
        byte[] oldData = (byte[])outBuffer.getData();
        byte[] newData = new byte[length];
        // Copy the previous data
        for (int i = 0; i < offset; i++) {
            newData[i] = oldData;
// Set the variables for the output buffer
outBuffer.setData(newData);
outBuffer.setLength(length);
outBuffer.setOffset(offset);
// Copy the data from in to out
int j = outBuffer.getOffset();
for (int i = inBuffer.getOffset(); i < inBuffer.getOffset() + inBuffer.getLength(); i++) {
((byte[])outBuffer.getData())[j++] = ((byte[])inBuffer.getData())[i];
return BUFFER_PROCESSED_OK;
That's untested code above, fix my dumb mistakes as necessary.

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