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
Similar Messages
-
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... -
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(); -
Using RTP stream as data source
I have successfully written a program that reads G.711 audio from a wav file and sends it out over the network.
I'm trying to modify that program so that instead of getting its source audio from a file, it instead receives the audio from an RTP stream. So, in effect, it is simply receiving audio from one socket and sending it back out another.
The error I'm encountering:
[java] javax.media.NoProcessorException: Cannot find a Processor for: rtp://172.30.18.140:32916
[java] at javax.media.Manager.createProcessorForContent(Manager.java:1663)
[java] at javax.media.Manager.createProcessor(Manager.java:627)
Here is the code fragment where the exception occurs:
processor = javax.media.Manager.createProcessor(new MediaLocator("rtp://172.30.18.140:32916"));
Can anybody help? Thanks!
<removed funky formatting><br>
Message was edited by:
fehHi,
I Had a very similar problem, but in addiction I had to store data in a buffer before retransmission. I resolved it employing low level classes. At this time, I still do not know any other solution, and nobody helped me in any way, also in this forum...
So, first you have to access single frames of the stream (Buffer or ExtBuffer objects) using a RawBufferParser, then you have to reconstruct a DataSource by multiplexing the frames (use an appropriate Multiplexer class, such as RTPSynchBufferMux).
The big problem is that there is no documentation avalaible about how to use these classes: I have lerned all I know by "reverse engineering" (lots of hours spent in reading very long annoying code). A little help can be given by the PlugIn Viewer of JMStudio, that tells you which classes are to be employed.
I have not tried to retransmit data "straightforward", without access single frames, because that was not my task. Maybe if you use RTPManager, it is possible to get a DataSource by the ReceiveStream object, and then to pass it to another RTPManager to create a SendStream.
I do not know any simpler way to do that, but this does not mean that it does not exist...!
good luck.
Alberto M. (Italy) -
Problem to create RTP stream on Linux by using JMF studio
Hello All,
I am trying to use JMStudio in order to create RTP stream
I have a problem with capture
I have run Jmfinit I got following messages
JavaSound Capture Supported = true
JavaSoundAuto: Committed ok
java.lang.Error: Can't open video card 0
java.lang.Error: Can't open video card 1
java.lang.Error: Can't open video card 2
java.lang.Error: Can't open video card 3
java.lang.Error: Can't open video card 4
java.lang.Error: Can't open video card 5
java.lang.Error: Can't open video card 6
java.lang.Error: Can't open video card 7
java.lang.Error: Can't open video card 8
java.lang.Error: Can't open video card 9
it seems to me that i have no directsoundcapture
therefore i can not create RTP package stream
How can i add direct sound capture to my capture devices.
is there any suggestion for this issue?
King regards
BEKIR BALCIK
ARGELA TECHNOLOGIESIn $ORACLE_BASE/admin you should have a directory structure for the viton1 instance : create a similar directory structure for the viton2 instance.
In the pfile subdirectory create the initviton2.ora file : you can copy it from initviton1.ora file and change all occurrences of viton1 to viton2.
At OS prompt : export ORACLE_SID=viton2, then
sqlplus /nolog
conn / as sysdba
startup nomount pfile=$ORACLE_BASE/admin/viton2/pfile/initviton2.ora
then CREATE DATABASE..... -
Hi guys,
what I need to do is to stream YUV images between two JMF applications. I am using VideoTransmit application (taken from this site) to transmit and JMStudio to receive. From a little research I have noticed the following to transmit:
1) Create the DataSource using the Manager.CreateDataSource(MediaLocator)
2) create processor using the above created datasource
3) create player etc.
My question now is the following:
How can I transmit a YUV file? How can I create the DataSource which has this YUV file without having problems for the further creation of processor etc?
Thanks in advance guys,
ChrisOk, I think I know what was confusing me. Apparently the setRate() isn't effecting the rate of stream. The audio file I am streaming runs about five minutes usually, so I sped the rate up to 10. It sped the process of writing to a file up significantly, but writing to an RTP stream seems to take the full run-time of the file. Does this sound right or is there something else I'm missing?
Thanks,
Khanathor -
Hello
I need help can someone tell me how i play and save at the same time an rtp stream?
ThanksThanks for your response.
Very much appreciated. Was very informative.
This is my current situation with 3 your suggestions:
Daniele,
Your suggestion 1’s result:
In Wireshark ----> Under Statistics --->I have VoIP calls.
(I don’t see VoIP calls under Telephony –> may be a different version of Wireshark).
Anyway, there is only one call because the Wireshark had a Capture Filter to track information between one source and one destination IP address. So I select that call and click on Player button and then click on Decode button. Then I select the forward stream (From IP1 to IP2) and click on play and I don’t hear anything at all. All silence. Same when I select the reverse stream from IP2 to IP1 and play.
Your suggestion 2’s result:
In Wireshark ---> Under Statistics ---> I Selected Stream Analysis (Did not select Show All Streams – not sure what the difference is) then ---> Save Payload ----> Select “au” instead of raw and it says – “Can’t save in a file:saving in au format supported only for alaw / ulaw stream
Your suggestion 3’s result:
Saved the file in .raw format. Opened Audacity and imported the file as raw and specified FIRST the A-Law codec for G.711A and selected 8000hz and that didn’t work and SECOND tried the u-Law coding for G.711u and selected the sample frequency again equal to 8000 Hz and that didn't work.
Didn't work means:
When I played the imported information I get all noise (like heavy metallic sound) and no voice.
So my guess is that this capture is neither A-Law or u-Law codec - right. This capture was given to me by a customer.
Any other suggestions – much appreciated Daniele. -
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,
Jarekbgl,
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 -
Mixing audio RTP-Streams to conference
I'm trying to create a telefon conference applikation with jmf.
My problem:
I need to mix multiple incomming RTP-Streams (Audio/G711_ulaw) together into one outgoing RTP-Stream.
If I use a MergingDataSource i can get one DataSource with multiple Tracks, but i'm not able to send them out together over one RTP-Session.
Is there any way to do this?
I have also thaught about using RawBufferMux or WAVMux to multiplex the tracks included in the MergingDataSource ,but i cant find out how to use them.
Or is there an easyer way to create an conference over RTP?
I need to connect to cisco ip-phones, so i can only use the g711_ulaw codec.
Is there anyone out who can help me?Im sorry, but I met this problem in past and find it impossible to resolve.
Im also thought about MergingDataSource, about Many2One class and more and more but nothing I could use. And nobody could answer this question in this forum half of year ago.
Oh, nearly forgot: it is possible to write merged streams to file, seems possible to write them into file and then to transmit it to net from file. But how to realize more than one Processor?.. -
To retransmit the rtp streams received
hi !
Program A transmits media data to another program B using RTPSessionMgr.What B has to do is send the recived streams to another receiver program C.This has to play the stream.That is I am trying to implement a client router server model.
So i maintained a session between A & B and also between B & C.Once NewReceiveStream event occurs at B,it retrieves the datasource from the stream and uses the createSendStream() method of the sessionmanager and sends to C.
My problem is that,C receives the audio and video streams.but never plays it.I get a pink screen for video and no audio.
can somebody help me out in pointing out my mistake or tell me a new strategy to implement this.
Actually,i have tried to use only datagram sockets on the router side.ie I created 2 sockets for B to receive data. and forwarded to C using 2 other sockets .And this did work.But the client does not know the sender details.I require to maintain the sender,receiver reports.So i went for the session manager(RTPManager /RTPSessionMgr).
kindly help.Hi all!
Nice to meet ya, it very exciting that I got a project which will integrating the JMF and JXTA, it got RTP streams from the JMF framework and then sending it to the JXTA framework ,and sent to any peer in some groups to visualizing it .So some aspects of which is similar to yours ,would you mind add me to your contact list of MSN(my MSN account is:[email protected]) -
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 AMHi, 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 :( -
Receiving Video RTP Stream (JMF) in JME ( MMAPI ) - URGENT !!!
Hi Folks...
I�m trying to develop an application that sends the images from a web cam connected to the computer, to the PDA that the images can be viewed by the user...
My code for the JMF RTP Video Stream is as follows.
Processor proc = null;
javax.media.protocol.DataSource ds = null;
TrackControl[] tc = null;
int y;
boolean encodingOk = false;
Vector<javax.media.protocol.DataSource> datasources = new Vector<javax.media.protocol.DataSource>();
for( int x = 0 ; x < camerasInfo.length ; x++ ){
try {
proc = Manager.createProcessor(camerasInfo[x].getLocator());
}catch (NoProcessorException e) { System.out.println("Erro ao int�nciar PROCESSOR: " + e); }
catch (IOException e) { System.out.println("Erro ao int�nciar PROCESSOR: " + e); }
proc.configure();
try {
Thread.sleep(2000);
} catch (InterruptedException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
proc.setContentDescriptor( new ContentDescriptor(ContentDescriptor.RAW_RTP) );
tc = proc.getTrackControls();
for( y = 0; y < tc.length ; y++ ){
if (!encodingOk && tc[y] instanceof FormatControl){
if( ( (FormatControl) tc[y] ).setFormat( new VideoFormat(VideoFormat.RGB) ) != null ){
tc[y].setEnabled(true);
}else{
encodingOk = true;
}else{
tc[y].setEnabled(false);
proc.realize();
try {
Thread.sleep(2000);
} catch (InterruptedException e1) {
e1.printStackTrace();
try{
ds = proc.getDataOutput();
}catch(NotRealizedError e){
System.out.println("ERRO ao realizar datasource: " + e);
}catch(ClassCastException e){
System.out.println("Erro ao realizar datasource: " + e);
datasources.add(ds);
System.out.println( ds.getLocator() );
encodingOk = false;
MediaLocator ml = new MediaLocator("rtp://10.1.1.100:99/video");
try {
DataSink datSink = Manager.createDataSink(ds , ml);
datSink.open();
datSink.start();
} catch (NoDataSinkException e) {
System.out.println("Erro ao instanciar DataSink: " + e );
} catch (SecurityException e) {
System.out.println("Erro ao iniciar DataSink: " + e);
} catch (IOException e) {
System.out.println("Erro ao iniciar DataSink: " + e);
}I�m not sure if this code is correctly... it is ?
So... the next part of the systems runs on the PDA..
The code that access this RTP Stream is as follows..
VideoControl c = null;
try {
player = Manager.createPlayer("rtp://10.1.1.100:99/video");
c = (VideoControl) player.getControl("VideoControl");
tela = (Item) c.initDisplayMode( GUIControl.USE_GUI_PRIMITIVE, null);
player.start();
append(tela);
} catch (IOException e) {
str.setText(e.toString());
append( str );
} catch (MediaException e) {
str.setText(e.toString());
append( str );
}So when the APP try to create a player for "rtp://10.1.1.100:99/video" an MediaException is throwed..
javax.microedition.media.MediaException: Unable to create Player for the locator: rtp://10.1.1.100:99/video
So... I don�t know what happen =/
The error is in the PDA module ? Or in the computer�s initialization off the RTP Video Streaming ?
I need finish this job at next week... so any help is usefull..
Waiting for answers
Rodrigo KerkhoffFirst of all: before going onto the j2me part, make sure the server works, before doing anything else! Apparently, it doesn't...
The MediaLocator is generally used to specify a kind of URL depicting where the data put into the datasink should go to. In your case, This cannot be just some URL where you want it to act as a rtps server. You'll need to implement that server yourself I guess. -
Adding Effect on an incoming H.263/RTP stream
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. -
How can I check to see if 802.1Q VLAN tagging information is being stripped from the RTP stream by the workstation's NIC?
Some network cards strip the dot1q headers, so your packet capture won't have this data. If you're not seeing dot1q headers please check the following link from wireshark:
http://wiki.wireshark.org/CaptureSetup/VLAN#head-81781716144f2855ab0aff2f8b752e95f2562efb
Apply the previous settings to the machine you're capturing with. -
Having multiple threads for receiving RTP streams
Hello,
Developing an audio conference server, I have come to think that if I manage to separate the different audioReceivers who receive the RTP streams the performance could improve.
At this moment I have the main program, so the main thread let�s say, who initializes a new audioRx object for each remote client.
Would separation of the different receivers into different threads improve the applications performance?
has anyone thought of this? or done something similar?
Thanks for your help.
bgli need help from about the RTP stream from the same port
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