How to stream audio/video to real Audio Player

Hello
I am planning on doing a project on streaming audio/video onto Real Audio Player . I am very new to JMF . I would appreciate some pointers on how to start off. I ahve read the documents but am not able to get started right .
I wanted to know the following :
1. Isnt the RTP streaming streaming for live audio or video ? If I want to stream something which I have stored in my local machine then how do i do it - do i need to use RTP ?
2. I undertsand that if i need to stream the audio/video file I need to convert it to a format the Real Audio Player supports. How do i go about that ?
Any help would be greatly appreciated!!!
Thanks in advance
Shailaja

JMF doesn't have any support for Real media. Real has gone open source with their Helix Community, so maybe you could find some useful information there: http://www.helixcommunity.org

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package heyram;
import java.util.logging.Level;
import java.util.logging.Logger;
import javax.media.control.*;
import javax.media.protocol.*;
import javax.media.format.*;
import java.io.IOException;
import java.util.Vector;
import java.net.InetAddress;
import java.awt.*;
import java.io.*;
import java.net.InetAddress;
import javax.media.*;
import javax.media.protocol.*;
import javax.media.protocol.DataSource;
import javax.media.format.*;
import javax.media.control.TrackControl;
import javax.media.control.QualityControl;
import javax.media.rtp.*;
import javax.media.rtp.rtcp.*;
import com.sun.media.rtp.*;
import java.lang.Thread.*;
public class HEYRAM{
Format[] format=new Format[2];
Vector[] devices=new Vector[2];
CaptureDeviceInfo[] di=new CaptureDeviceInfo[2];
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Processor p=null;
Processor p1=null;
boolean result;
public HEYRAM(String address,int port,int ttl){
    try{
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                                                         port,
                                                         destaddr,
                                                         port + 1);
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                new SourceDescription(SourceDescription
                                      .SOURCE_DESC_EMAIL,
                                      "[email protected]",
                                                                 1,
                                      false),
                new SourceDescription(SourceDescription
                                      .SOURCE_DESC_CNAME,
                                      cname,
                                      1,
                                      false),
                new SourceDescription(SourceDescription
                                      .SOURCE_DESC_TOOL,
                                      "JMF RTP Player v2.0",
                                      1,
                                      false)
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                            userdesclist,
                            0.05,
                            0.25);
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             di[1] = (CaptureDeviceInfo)devices[1].elementAt(0);
              System.out.println(di[1].toString());
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            // exit if we could not find the relevant capture device.
         System.out.println("1234jjfjsajfjasf1");
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        try {
            p1 = Manager.createProcessor(di[1].getLocator());
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            System.out.println("1234jjfjsajfjasf3");
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       // p1.configure();
     result = waitForState(p1, Processor.Configured);
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     System.out.println("Couldn't realize processor");
        p1.setContentDescriptor(new ContentDescriptor( ContentDescriptor.RAW_RTP));
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        TrackControl track[] = p1.getTrackControls();
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        for (int i = 0; i < track.length; i++) {
            if (!encodingOk && track[i] instanceof FormatControl) {
                if (((FormatControl)track).
setFormat( new VideoFormat(VideoFormat.YUV)) == null) {
track[i].setEnabled(false);
else {
encodingOk = true;
else {
// we could not set this track to gsm, so disable it
track[i].setEnabled(false);
// Realize it and block until it is realized.
p1.realize();
result = waitForState(p1, Processor.Realized);
     if (result == false)
     System.out.println("Couldn't realize processor");
// block until realized.
// get the output datasource of the processor and exit
// if we fail
DataSource ds = null;
try {
ds = p1.getDataOutput();
} catch (NotRealizedError e){
//System.exit(-1);
System.out.println("1234jjfjsajfjasf4");
// Create a SessionManager and hand over the
// datasource for SendStream creation.
// The session manager then needs to be initialized and started:
// rtpsm.initSession(...);
// rtpsm.startSession(...);
try {
(rtpsm.createSendStream(ds, 0)).start();
} catch (IOException e){
System.out.println("1234jjfjsajfjasf6");
e.printStackTrace();
} catch( UnsupportedFormatException e) {
System.out.println("1234jjfjsajfjasf7");
e.printStackTrace();
public void createAudioSession(){
format[0] = new AudioFormat("linear",8000,8,1);
devices[0]= CaptureDeviceManager.getDeviceList( format[0]);

[SOLVED] How to stream audio output over wifi?

Hello,
I have following problem. I would like to listen to the music from my laptop on my home Onkyo AV receiver (amplifier). Both laptop and receiver are connected on the same network, laptop via wifi, receiver using LAN. Receiver is capable to play internet radio streams in various format - e.g. ogg or mp3.
My idea is to produce ogg stream on laptop and listen to this stream through amplifier. I was searching for this solution a found something that has always some disadvantage, like MPD + MPC (can play only from database of local files, not from jamendo or other network services), DLNA can play only local files, Icecast + Ices can play from predefined playlist.
My idea is to have something independent - something what takes audio data normally played on soundcard and feeds it to network stream, something what can work with any player - audio or video, or even with desktop notifications.
Does anybody here know how to do it?
Thanks!
Last edited by KejPi (2012-03-14 20:18:28)

Thank all of you for your hints. Finally I have decided for ALSA based solution that simply works for all applications smoothly in background and it is here when I need it - in other words, it is something that doesn't borther me when I do not need it. I use ALSA loopback with ffmpeg and ffserver for streaming in mp3 format.This is my final solution:
You need following:
alsa
ffmpeg
if you use KDE4, then you need phonon-gstreamer backend, as in my case phonon-vlc didn't work (it was heavily distorted)
Procedure:
Modprobe snd-aloop (2 loopback channels are enough)
modprobe snd-aloop pcm_substreams=2
For permanent solution follow arch wiki https://wiki.archlinux.org/index.php/Modprobe
Create ALSA configuration file and store it either to ~/.asoundrc (just for user) or to /etc/asound.conf as system wide:
pcm.!default {
type asym
playback.pcm "LoopAndReal"
capture.pcm "hw:0,0"
hint {
show on
description "Default with loopback"
#"type plug" is mandatory to convert sample type
pcm.LoopAndReal {
type plug
slave.pcm mdev
route_policy "duplicate"
hint {
show on
description "LoopAndReal"
pcm.mdev {
type multi
slaves.a.pcm pcm.MixReal
slaves.a.channels 2
slaves.b.pcm pcm.MixLoopback
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
pcm.MixReal {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
#rate 48000
#rate 44100
#periods 128
#period_time 0
#period_size 1024 # must be power of 2
#buffer_size 8192
pcm.MixLoopback {
type dmix
ipc_key 1025
slave {
pcm "hw:Loopback,0,0"
#rate 48000
#rate 44100
#periods 128
#period_time 0
#period_size 1024 # must be power of 2
#buffer_size 8192
You can play with sample rates and buffer sizes you you have any problem. This configuration works on my system.
Prepare ffserver configration and store either to default location /etc/ffserver.conf as system wide setup or anywhere to your home:
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
NoDaemon
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<Feed feed1.ffm>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
#ACL allow 127.0.0.1
</Feed>
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
# MP3 audio
<Stream stream.mp3>
Feed feed1.ffm
Format mp2
AudioCodec libmp3lame
AudioBitRate 320
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Format ogg
#AudioCodec libvorbis
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.1.0 192.168.1.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>
This sets ffserver for streaming in MP3 format, stereo, 320kbps. Unfortunately I haven't succeeded with OGG Vorbis streaming.
Now you have all configuration you need and if you want to stream following two commands do that:
ffserver -f ffserver.conf
ffmpeg -f alsa -ac 2 -i hw:Loopback,1,0 http://localhost:8090/feed1.ffm
You can test it for example by mplayer:
mplayer http://YourLinuxBox:8090/stream.mp3
And that's it. Sound is played by normal sound card and sent to stream simultaneously. If you do not want to listen sound from computer you can mute your soundcard. It has an advantage that one can normally listen to music on the computer with or without streaming and in both cases without any reconfiguration. To start streaming just call ffserver and ffmpeg.
Advantages:
+ very simple solution without any special sound server
+ no special SW required (in my case I had already instaled all I need for that)
+ streaming on request by two simple commands
+ normal soundcard function
+ streaming in MP3 format that is supported by many home AV receivers
Disadvantages
- phonon-vlc backend not compatible (also VLC does not work)
- OGG streaming does not work
- some latency (~ 5 sec)
- all sounds are sent to stream, including various desktop notifications (in KDE could be managed by phonon)

How to stream audio w/o connecting AX as gateway?

Have been struggling with this one for a while – would appreciate any help you might offer.
I have my MAC connected to my university network from my room. The Univ is fairly strict with network things and so will not allow students to run wireless networks (otherwise I'd happily plug in my Airport Express as the internet gateway and connect my Powerbook through that – we have to give our IT dept the MAC address of our computers and we get assigned a fixed IP address).
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Thanks in advance
PowerBook G4 Mac OS X (10.4.9)

"Is there some sort of "standalone" mode for AX that allows me to use it just as a receiver of the audio stream from iTunes?"
Welcome to Discussions, rnorum!
Yes, but that's still setting up a network between your Mac and your Express/stereo connection. Your AirPort Express does have its own MAC address - its AirPort ID. Every wireless device has a MAC address, even non-Macs have one. If the Univ is fairly strict and maintains the integrity of its network, they'll be able to easily discover any network you setup.
If you turn on your AirPort and look in your menu bar's AirPort icon and can see no other networks, then it's safe to assume you shouldn't try this - as others will be able to easily discover it. If instead you can see that everyone else has an iTunes network, then ask around to see if they just did it, or went through the proper procedure.
Assuming that you've already cleared the Express's old profiles by resetting to factory default
[refer to the last section "How to perform a factory default reset" on this page: http://docs.info.apple.com/article.html?artnum=108044 ]
you should see the Express's default name (Apple Network ____) in your menu bar's AirPort icon. If you've already connected your stereo to the Express with the proper audio cabling, use the utility "AirPort Setup Assistant" to setup that network. It's wise to give the connection its own name rather than to leave it set on the default network name. The Assistant should see your Express's stereo connection, if not, try again, don't continue on and setup a bogus internet connection. If you do that, you'll just have to again reset the Express:)

Streaming audio player problems.

I would be grateful if someone could help me with the following problems.
I have developed an audio streaming app that relies on OSMF framework in Android and StageVideo / Netstream classes in iOS (OSMF is HLS agnostic).
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When the screen goes dark, the app takes almost a minute to switch between streams. When testing in AIR simulator and when the device screen is not dark, the app moves to the next track quickly. Why? Is this related to the fact that the framerate slows to 4 per second when device screen is dark?
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Just to be clear, using the same sound file Ableton Live will crash but other media players play it back fine? If that's the case, I would contact Ableton support.
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How to stream external video thru iTunes?

Hey Apple community!
So I've got a bit of a question. Is there a way to stream an internet video (such as from youtube) thru iTunes onto my computer? What I'm saying, for example, is I wanna watch a video on YouTube, but I wanna watch it in an iTunes movie window. The reason being that I absolutely love the fact that iTunes allows you to play your movies and what-not over every other window on the screen if you want. I use this feature every day as I use my computer for video editing purposes (among others) but oftentimes I have to look up a tutorial video on how to do something. It would be immensly helpful if I could keep the tutorial video over my other windows (thru iTunes, unless theres another way to do this) so it's easier to follow along.
Thanks for the help!

Maybe try a YouTube downloader.. no way to stream that I know of.

How to stream audio/video to real Audio Player

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