Something to play Real Audio files? (.ra)
I sometimes see Real Audio files on websites, on Windows, they have RealPlayer that will play them. Is there a similar app on linux/Arch. I'm mostly interested in Audio (*.ra), though I am also curious about the video files (*.ram).
I'm hoping to find a plug-in for mozilla-firefox, or link in an outside app.
I've been searching through the package database, but didnt see anything.
thx
Hrmmm ... perhaps the problem is with 'mplayer-plugin', 'cause when I do an 'about:plugins' in FireFox I don't see any entries for '.rm', '.rpm', etc.:
QuickTime Plug-in 6.0, Windows Media Player Plugin are supported by mplayerplug-in
File name: mplayerplug-in.so
mplayerplug-in 2.60
Video Player Plug-in for QuickTime, RealPlayer and Windows Media Player streams using MPlayer
JavaScript Enabled
MIME Type Description Suffixes Enabled
video/quicktime Quicktime mov Yes
video/x-quicktime Quicktime mov Yes
image/x-quicktime Quicktime mov Yes
video/quicktime Quicktime mp4 Yes
video/quicktime Quicktime - Session Description Protocol sdp Yes
application/x-quicktimeplayer Quicktime mov Yes
video/x-ms-asf-plugin Windows Media asf,asx Yes
video/x-msvideo AVI avi Yes
video/msvideo AVI avi Yes
application/x-mplayer2 WMV wmv Yes
video/x-ms-wm MSNBCPlayer asf Yes
video/x-ms-asf Windows Media asf,asx Yes
video/x-ms-wmv Windows Media wmv Yes
video/x-ms-wmp Windows Media wmp Yes
video/x-ms-wvx Windows Media wvx Yes
audio/x-ms-wax Windows Media wax Yes
audio/x-ms-wma Windows Media wma Yes
application/x-drm-v2 Windows Media asx Yes
audio/wav Microsoft wave file wav Yes
audio/x-wav Microsoft wave file wav Yes
video/mpeg MPEG mpg,mpeg Yes
audio/mpeg MPEG mpg,mpeg Yes
video/x-mpeg MPEG mpg,mpeg Yes
video/x-mpeg2 MPEG2 mpv2,mp2ve Yes
audio/mpeg MPEG mpg,mpeg Yes
audio/x-mpeg MPEG mpg,mpeg Yes
audio/mpeg2 MPEG audio mp2 Yes
audio/x-mpeg2 MPEG audio mp2 Yes
audio/mpeg3 MPEG audio mp3 Yes
audio/x-mpeg3 MPEG audio mp3 Yes
audio/mp3 MPEG audio mp3 Yes
video/mp4 MPEG 4 Video mp4 Yes
video/fli FLI animation fli,flc Yes
video/x-fli FLI animation fli,flc Yes
video/vnd.vivo VivoActive viv,vivo Yes
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Playing an audio file from applet
Hi friends,
I have a problem to be solved .
Please help me.
I have to play an audio file from applet.When the applet loads it shold start playing .There is a progress bar in the applet it shold alsomove corresponding to that.
How can i do that.Give me an idea.
I know nothing about jmf.
I am using j2sdk1.4,windows 2000 server.Andrew,
Forgive my naivety, but i struggle with JAVA! I am desperately trying to create a JAVA plugin to SERVOY and have to date been successful with a few things. Record, Playback, convert to .spx.
I am really struggling with implementing a rewind and fastforward function of the type you have. I thought it would be easy, but it appears not.
At the bottom is the complete code from my class that is called by the servoy plugin wrapper,
i had thought that adding some simple jump 500 ms would have been easy in something like this:
public AudioStream js_FastForward (AudioStream as) throws IOException {
!!! Line or two here to move the play head forward !!!
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return as;
Can you give me any pointers to how i code that fast forward bit.
many thanks
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package com.d2e.MyPlugin;
import com.servoy.j2db.scripting.IScriptObject;
import javax.swing.*;
import java.awt.*;
import java.awt.event.*;
import java.io.*;
import java.awt.Button;
import javax.sound.sampled.*;
import org.xiph.speex.*;
import org.xiph.speex.spi.*;
import sun.audio.*; //import the sun.audio package
public class MyPluginProvider implements IScriptObject {
public void JSencode()
throws IOException{
/** Version of the Speex Encoder */
final String VERSION = "Java Speex Command Line Encoder v0.9.7 ($Revision: 1.5 $)";
/** Copyright display String */
final String COPYRIGHT = "Copyright (C) 2002-2004 Wimba S.A.";
/** Print level for messages : Print debug information */
final int DEBUG = 0;
/** Print level for messages : Print basic information */
final int INFO = 1;
/** Print level for messages : Print only warnings and errors */
final int WARN = 2;
/** Print level for messages : Print only errors */
final int ERROR = 3;
int printlevel = INFO;
/** File format for input or output audio file: Raw */
final int FILE_FORMAT_RAW = 0;
/** File format for input or output audio file: Ogg */
final int FILE_FORMAT_OGG = 1;
/** File format for input or output audio file: Wave */
final int FILE_FORMAT_WAVE = 2;
int srcFormat = FILE_FORMAT_OGG;
int destFormat = FILE_FORMAT_WAVE;
int mode = -1;
int quality = 4;
/** Defines the encoders algorithmic complexity. */
int complexity = 3;
/** Defines the number of frames per speex packet. */
int nframes = 1;
/** Defines the desired bitrate for the encoded audio. */
int bitrate = -1;
/** Defines the sampling rate of the audio input. */
int sampleRate = -1;
/** Defines the number of channels of the audio input (1=mono, 2=stereo). */
int channels = 1;
/** Defines the encoder VBR quality setting (float from 0 to 10). */
float vbr_quality = -1;
/** Defines whether or not to use VBR (Variable Bit Rate). */
boolean vbr = false;
/** Defines whether or not to use VAD (Voice Activity Detection). */
boolean vad = false;
/** Defines whether or not to use DTX (Discontinuous Transmission). */
boolean dtx = false;
//HArd code src format and dest
// private String srcPath ="junk.wav";
srcFormat = FILE_FORMAT_WAVE;
destFormat = FILE_FORMAT_OGG;
//destPath="junk.spx";
byte[] temp = new byte[2560]; // stereo UWB requires one to read 2560b
final int HEADERSIZE = 8;
final String RIFF = "RIFF";
final String WAVE = "WAVE";
final String FORMAT = "fmt ";
final String DATA = "data";
final int WAVE_FORMAT_PCM = 0x0001;
// Open the input stream
DataInputStream dis = new DataInputStream(new FileInputStream("junk.wav"));
// Prepare input stream
//DP - Sort out the Wave File
// read the WAVE header
dis.readFully(temp, 0, HEADERSIZE+4);
// Read other header chunks
dis.readFully(temp, 0, HEADERSIZE);
String chunk = new String(temp, 0, 4);
int size = readInt(temp, 4);
while (!chunk.equals(DATA)) {
dis.readFully(temp, 0, size);
if (chunk.equals(FORMAT)) {
typedef struct waveformat_extended_tag {
WORD wFormatTag; // format type
WORD nChannels; // number of channels (i.e. mono, stereo...)
DWORD nSamplesPerSec; // sample rate
DWORD nAvgBytesPerSec; // for buffer estimation
WORD nBlockAlign; // block size of data
WORD wBitsPerSample; // Number of bits per sample of mono data
WORD cbSize; // The count in bytes of the extra size
} WAVEFORMATEX;
if (readShort(temp, 0) != WAVE_FORMAT_PCM) {
System.err.println("Not a PCM file");
return;
channels = readShort(temp, 2);
sampleRate = readInt(temp, 4);
if (readShort(temp, 14) != 16) {
System.err.println("Not a 16 bit file " + readShort(temp, 18));
return;
// Display audio info
if (printlevel <= DEBUG) {
System.out.println("File Format: PCM wave");
System.out.println("Sample Rate: " + sampleRate);
System.out.println("Channels: " + channels);
dis.readFully(temp, 0, HEADERSIZE);
chunk = new String(temp, 0, 4);
size = readInt(temp, 4);
if (printlevel <= DEBUG) System.out.println("Data size: " + size);
//DP ENd sort wave file
//Now Choose the mode , we have a file sampled at 44100
mode = 2; // Ultra-wideband
// Construct a new encoder
SpeexEncoder speexEncoder = new SpeexEncoder();
speexEncoder.init(mode, quality, sampleRate, channels);
if (complexity > 0) {
speexEncoder.getEncoder().setComplexity(complexity);
if (bitrate > 0) {
speexEncoder.getEncoder().setBitRate(bitrate);
if (vbr) {
speexEncoder.getEncoder().setVbr(vbr);
if (vbr_quality > 0) {
speexEncoder.getEncoder().setVbrQuality(vbr_quality);
if (vad) {
speexEncoder.getEncoder().setVad(vad);
if (dtx) {
speexEncoder.getEncoder().setDtx(dtx);
// Display info
// Open the file writer
AudioFileWriter writer;
if (destFormat == FILE_FORMAT_OGG) {
writer = new OggSpeexWriter(mode, sampleRate, channels, nframes, vbr);
else if (destFormat == FILE_FORMAT_WAVE) {
nframes = PcmWaveWriter.WAVE_FRAME_SIZES[mode-1][channels-1][quality];
writer = new PcmWaveWriter(mode, quality, sampleRate, channels, nframes, vbr);
else {
writer = new RawWriter();
writer.open("junk.spx");
writer.writeHeader("Encoded with: " + VERSION);
int pcmPacketSize = 2 * channels * speexEncoder.getFrameSize();
try {
// read until we get to EOF
while (true) {
dis.readFully(temp, 0, nframes*pcmPacketSize);
for (int i=0; i<nframes; i++)
speexEncoder.processData(temp, i*pcmPacketSize, pcmPacketSize);
int encsize = speexEncoder.getProcessedData(temp, 0);
if (encsize > 0) {
writer.writePacket(temp, 0, encsize);
catch (EOFException e) {}
writer.close();
dis.close();
* Converts Little Endian (Windows) bytes to an int (Java uses Big Endian).
* @param data the data to read.
* @param offset the offset from which to start reading.
* @return the integer value of the reassembled bytes.
protected static int readInt(final byte[] data, final int offset)
return (data[offset] & 0xff) |
((data[offset+1] & 0xff) << 8) |
((data[offset+2] & 0xff) << 16) |
(data[offset+3] << 24); // no 0xff on the last one to keep the sign
* Converts Little Endian (Windows) bytes to an short (Java uses Big Endian).
* @param data the data to read.
* @param offset the offset from which to start reading.
* @return the integer value of the reassembled bytes.
protected static int readShort(final byte[] data, final int offset)
return (data[offset] & 0xff) |
(data[offset+1] << 8); // no 0xff on the last one to keep the sign
AudioFormat audioFormat;
TargetDataLine targetDataLine;
public Class[] getAllReturnedTypes() {
// TODO Auto-generated method stub
return null;
public String[] getParameterNames(String arg0) {
// TODO Auto-generated method stub
return null;
public String getSample(String arg0) {
// TODO Auto-generated method stub
return null;
public String getToolTip(String arg0) {
// TODO Auto-generated method stub
return null;
public boolean isDeprecated(String arg0) {
// TODO Auto-generated method stub
return false;
public String js_Record (String name){
captureAudio();
return "Started Recording " +name;
public String js_StopRecord (String name) throws IOException{
//new ActionListener(){
// public void actionPerformed(ActionEvent e)
//Terminate the capturing of input data
// from the microphone.
targetDataLine.stop();
targetDataLine.close();
// }//end actionPerformed
//};//end ActionListener
// JSpeexEnc ("junk.wav","output.spx");
JSencode();
return "Stop Records " +name;
//Play audio file
public AudioStream js_Playback (String name) throws IOException{
InputStream in = new FileInputStream("junk.wav");
AudioStream as = new AudioStream(in);
AudioPlayer.player.start(as);
return as;
public AudioStream js_ContPlay (AudioStream as) throws IOException {
AudioPlayer.player.start(as);
return as;
//Stop Play
public AudioStream js_Stop_Playback (AudioStream as) throws IOException{
AudioPlayer.player.stop(as);
return as;
//This method captures audio input from a
// microphone and saves it in an audio file.
private void captureAudio(){
try{
//Get things set up for capture
audioFormat = getAudioFormat();
DataLine.Info dataLineInfo =
new DataLine.Info(
TargetDataLine.class,
audioFormat);
targetDataLine = (TargetDataLine)
AudioSystem.getLine(dataLineInfo);
//Create a thread to capture the microphone
// data into an audio file and start the
// thread running. It will run until the
// Stop button is clicked. This method
// will return after starting the thread.
new CaptureThread().start();
}catch (Exception e) {
e.printStackTrace();
System.exit(0);
}//end catch
}//end captureAudio method
// This method creates and returns an
// AudioFormat object for a given set of format
// parameters. If these parameters don't work
// well for you, try some of the other
// allowable parameter values, which are shown
// in comments following the declarations.
private AudioFormat getAudioFormat(){
float sampleRate = 44100.0F;
//8000,11025,16000,22050,44100
int sampleSizeInBits = 16;
//8,16
int channels = 1;
//1,2
boolean signed = true;
//true,false
boolean bigEndian = true;
//true,false
return new AudioFormat(sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}//end getAudioFormat
class CaptureThread extends Thread{
public void run(){
AudioFileFormat.Type fileType = null;
File audioFile = null;
//Set the file type and the file extension
// based on the selected radio button.
//if(aifcBtn.isSelected()){
// fileType = AudioFileFormat.Type.AIFC;
// audioFile = new File("junk.aifc");
// }else if(aiffBtn.isSelected()){
// fileType = AudioFileFormat.Type.AIFF;
// audioFile = new File("junk.aif");
// }else if(auBtn.isSelected()){
// fileType = AudioFileFormat.Type.AU;
// audioFile = new File("junk.au");
// }else if(sndBtn.isSelected()){
// fileType = AudioFileFormat.Type.SND;
// audioFile = new File("junk.snd");
// }else if(waveBtn.isSelected()){
fileType = AudioFileFormat.Type.WAVE;
audioFile = new File("junk.wav");
// }//end if
try{
targetDataLine.open(audioFormat);
targetDataLine.start();
AudioSystem.write(
new AudioInputStream(targetDataLine),
fileType,
audioFile);
}catch (Exception e){
e.printStackTrace();
}//end catch
}//end run
}//end inner class CaptureThread
} -
Is it possible to play two audio files simultaneously.
Hi,
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I modified the 'SpeakHere' to play two audio files simultaneously, it worked perfectly on simulator but get hanged on iphone.
To this I implemented NSTimer, which regularly checks for both player instance every second.As soon as any player gets stopped Timer task stops the second one forcefully and again starts both.
thanks.Hi Jay
Do you have the eLearning Suite? If so, you have Adobe Presenter available to you. If you don't, you should be able to download the Presenter installation file and evaluate it.
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Just musing out loud... Rick
Helpful and Handy Links
Captivate Wish Form/Bug Reporting Form
Adobe Certified Captivate Training
SorcerStone Blog
Captivate eBooks -
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Hi All, i want to play a audio file from the point of mark.
i have used mark() method to mark but when i play the audio it is still playing the audio from the starting point. please help me.Hi,
Question:
1. You have a folder for each DVD whose video contents you edited.
Answer: Yes
Question:
You talk about projects 34, 35, 36, 37. Are these projects associated with only one specific DVD?
What is supposed to be in each project - just edited VTS_01_1.VOB or edits from the others in the series
such as VTS_01_2.VOB, VTS_01_3.VOB, etc.
Answer:
Project 34 has edited Premiere Elements file 34 and the VOB files (VST_01_1.VOB, VTS_01_2.VOB, etc) from DVD#34
Project 35 has edited Premiere Elements file 35 and the VOB files (VST_01_1.VOB, VTS_01_2.VOB, etc) from DVD#35
Project 36 has edited Premiere Elements file 36 and the VOB files (VST_01_1.VOB, VTS_01_2.VOB, etc) from DVD#36
etc.
Question:
2. If you screen each DVD video file VOB for a given DVD disc, does the content follow sequentially and cleanly?
Answer:
Yes
When I run the VST_01_1.VOB, VTS_01_2.VOB, etc from each project the audio and the video are clean and match.
When I run the edited Premiere Elements file that were created after project 34 edited Premiere Elements file project 35, 36, 37 etc. I am getting the audio from project 34
When I run the edited Premiere Elements file that were created before project 34 edited Premiere Elements file project 1, 2, all the way to 33. I am getting the correct audio
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Ron -
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I would like to play an audio file (edc.caf) that is located in this directory....
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[code]
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[code]
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But i do not hear any audio out of the iPod touch.
I've tried two different methods of playing the audio,
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Can anyone confirm the location is correct???
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Help me play an audio file in my GUI
Hi
I've been having some trouble getting code to play an audio file in my GUI
Basically all I want to do is play a file in the same directory as the program, and have it loop. No controls, no buttons to change the play state.
Could someone point me to or help me implement this with some example code? I'm really stuck and I'm just not sure how to implement it into the existing GUI class.
Thanks for any helpHi
I've done so, and the method gets called without any problems, but doesnt play anything.
Have a look -
public void audioFile()
try {
// From file
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AudioFormat.Encoding.PCM_SIGNED,
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format.getChannels(),
format.getFrameSize()*2,
format.getFrameRate(),
true); // big endian
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Clip.class, stream.getFormat(), ((int)stream.getFrameLength()*format.getFrameSize()));
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clip.open(stream);
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} catch (MalformedURLException e) {
} catch (IOException e) {
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Hi all,
I am using a myDAQ (first time). I am trying to play an audio file at a certain time in my VI.
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Go to Solution.Why no just play a WAV file? Use Play Sound File.vi as shown here.
Richard -
Hello everybody,
I'm trying to play an audio file over HTTP protocol. The code is the following:
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static String url = "http://localhost/audio/Reklam1.wav";
static String urlFile = "file:///C://tmp/audio/Reklam1.wav";
* @param args
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// TODO Auto-generated method stub
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dataS.connect();
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// TODO Auto-generated catch block
e.printStackTrace();
} catch (IOException e) {
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e.printStackTrace();
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// TODO Auto-generated catch block
e.printStackTrace();
}Now, if I'm trying to play the file from the local disk (+urlFile+ - using file protocol), everything goes well, but if I'm trying to play the same file from the network (using http protocol) I get the following exception:
javax.media.NoPlayerException: Cannot find a Player for: javax.media.protocol.URLDataSource@fa9cf
Can somebody tell me what I'm doing wrong?
Thank you!Ah',..okay. I constructed the DataSource like follows:
Buffer mediaBuffer = new Buffer();;
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for (;;) {
int data = bufIn.read();
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break;
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System.out.println(mediaBuffer.getLength());
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httpStream[0].read(mediaBuffer);
ds.connect();
ds.start();
Player player = Manager.createPlayer(ds);
player.start();But I still cannot play an audio file (wav format) over the HTTP. Application starts but nothing happened.
I make the modification like you suggested. (into the DataSource, more exactly for method getStreams()).
I renamed as well the HttpDatasource into HttpStream.
Did you actually tried the code? I've been reading the instruction how to test the code but I did not be able to run the example. -
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How do I play an audio file as a soundtrack for entire keynote?
https://developer.apple.com/mac/library/samplecode/PlayFile/Introduction/Intro.h tml
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Is possible to play .amr audio files on Nokia Lumi...
Is possible to play .amr audio files on Nokia Lumia 800? Thank you.
Hi,
The Lumia, being a Windows Phone, supports these audio formats;
AAC, AMR-NB, AMR-WB, HE-AAC v1, HE-AAC v2, M4A, MP3, WAV, WMA 10 Pro, WMA 9
So yes, AMR is supported.
Press the 'Accept As Solution' icon if I have solved your problem, click on the Star Icon below if my advice has helped you! -
How do i play multiple audio files simultaneously using AudioQueue?
I've basically gutted what I don't need from the "Speakhere" sample app and I'm left with something I can use to play audio files. I converted wav files to aiff and then renamed the aiff to caf (Just in case this matters). I even tried shortening the files to 10 seconds long, but no difference. I am able to play files singly, but am unable to play more than one file at the same time. When I try to play multiple files, the first one plays correctly but the second will play a horrible long beeping sound. Sounds almost like a buzzer and sometimes will sound like a high pitched beep. My code has all the audio components like the speakhere app (AudioQueueObject, AudioPlayer, AudioViewController). I've changed the AudioViewController to be just my "AudioController" class where when I initialize, I pass in the name of the file and it replaces this line of code:
CFStringRef fileString = (CFStringRef) [NSString stringWithFormat: @"%@/Recording.caf", self.recordingDirectory];
with this:
CFStringRef fileString = (CFStringRef) [NSString stringWithFormat: fileName, self.recordingDirectory];
Any suggestions? I've been yanking my hair out trying to figure this out and I simply can't. I'm about ready to pull an "Office Space" but instead of beating up a printer, I'm going to destroy my macbook.
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MyAudioManager *manager1 = [[MyAudioManager alloc] initWithFile:@"file1.caf"];
[manager1 play];
MyAudioManager *manager2 = [[MyAudioManager alloc] initWithFile:@"file2.caf"];
[manager2 play];
// etc...
If you want to play short files (~10s or less), you can use the simpler AudioServices functions along with SystemSoundID's. It will work fine if you have playback from AudioQueue and AudioServices at the same time. -
Playing an audio file in the mainBundle through an NSDictionary
I have an audio file audio1.caf in my mainBundle.
In the mainViewController I want to use an NSDictionary with audio1.caf being the object for the key audio1Key
In my detailViewController I want to have the user tap a button and the audio plays.
This sounds like a very simple thing but search as I might I can't seem to find the answer.
Here is the code for my dictionary in the mainViewController
if ([vc isKindOfClass:[detailViewController class]])
detailViewController*lcVC =(detailViewController*)vc;
NSMutableDictionary *mydDictionary = [NSMutableDictionary dictionaryWithObjectsAndKeys:
@"audio1.caf",@"firstAudioKey",nil];
lcVC.myDictionary = myDictionary;
[self.navigationController pushViewController:vc animated:YES];
Then in my detailViewController I use
-(IBAction)buttonPressedid)sender {
NSError *error = nil;
NSString *firstAudio = [myDictionary objectForKey:@"firstAudioKey"];
NSURL *url = [NSURL fileURLWithPath:firstAudio];
theAudio = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error];
theAudio.delegate =self;
theAudio.volume = 1.0;
theAudio.numberOfLoops = 0;
NSLog (@"%@",url);
if ([theAudio isPlaying]){[theAudio pause]; }else{[theAudio play];}}
So I know things are getting passed by the dictionary ok because other things are showing up (text, and other audio) stored in the documents directory.
However this shows that the url path is audio1.caf --file://localhost/ So I am gettin the name of the objectForKey but not the whole path. I think I need to look in the mainBundle and get the path to the file but I cannot seem to be able to figure this out. I have looked all over the internet for a discussion, or tutorial. but no luck. Can someone point out where I'm going wrong or suggest where to look. ThanksI finally figured this out DUH! Here's my code if anyone wants to look at it and tell me if I'm doing something wrong.
-(IBAction)buttonPressed2:(id)sender {
NSError *error = nil;
NSString *someAudio = [[NSBundle mainBundle]bundlePath];
NSString *audioPath = [someAudio stringByAppendingPathComponent:[myDictionary objectForKey:@"firstAudioKey"]];
NSURL *url = [NSURL fileURLWithPath:audioPath];
theAudio7 = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error];
theAudio7.delegate =self;
theAudio7.volume = 10.0;
theAudio7.numberOfLoops = 0;
NSLog (@"%@",firstAudioKey);
if ([theAudio7 isPlaying]){[theAudio7 pause]; }else{[theAudio7 play];}}
Thanks to everyone. -
How to play an audio file from the begin.........
Hello to everybody,
I am working on a application that is supposed to manage audio file. At the moment I 'm using the SourceDataLine instead of the Clip. But I've got a problem. When the user pushes the Stop button I want that if he pushes Play again the track starts from the begin. I've used the drain() and flush() method but it doesn't work because the track resumes from the point when has been interrupted. Here part of the code:
@Override
public void run()
try
if(dataline!=null)
dataline.open(audio_format);
dataline.start();
int i=0;
while((i=audio_input_stream.read(buffer, 0, buffer.length))!=-1 && flag==false)
if(i>0)
dataline.write(buffer, 0, i);
dataline.drain();
dataline.stop();
dataline.flush();
dataline=null;
dataline.close();
And also I'd like to know how to navigate through the audio file by a slider or something like that, so how much time is passed.
Thank you in advance.
Maurizio Di Vittomauriziodvt wrote:
I've tried to use it but I found out that if I use quite large wav files an exception is thrown, Aha! I expected as much. That is why I developed BigClip.
..so I thought to change to SourceDataLine even if Clip has the right stuff to manage my audio files. So what do you suggest? Is there a way to avoid such problem? I mean, how can I be sure that I'm able to use every wav files? Not every WAV file can be opened using SourceDataLine, but if SDL can open it, so can BigClip. -
Cannot play Real Audio stream in my N82
Hi there,
i have this Real Audio link:
rtsp://stream2.rbb-online.de/encoder/radioeins-live.ra
It plays fine on my desktop computer. But, when I use the URL in the web browser on my N82, Realplayer opens, tries to establish a connection and thats it... nothing....
I have correct settings, I connect via WLAN and have opened the UDP ports on my router, so no issues here. I would be grateful for any hint!unten auf deutsch
No solution found yet. I believe the radio station's stream is encoded in a real audio format the installed Realplayer does not understand. This also applies to MPEG4 AVC1 encoded video podcast streams that cannot be handled by the Realplayer used on Nokia devices. More and more podcasts use this codec.I hope, Nokia will adress these issues, the Realplayer on the N82 and maybe on other devices as well is dated 2007 and obviously can only handle very basic codecs. Nokia, do something about this! The only way out *may* be the N900 which *might* have much more powerful media players due to its Linux based OS.
Ich vermute, das ist ein real audio Format, dass der Realplayer nicht beherrscht. Bei Videopodcasts im mpeg4 avc1 Format ist das auch so, den Videopodcast z.B. von "neues" oder "Neues aus der Anstalt" kann man auf vielen Nokia auch nicht wiedergeben.Keine Ahnung, ob es für Mobilgeräte einen neueren Realplayer gibt, der damit klarkommt, der installierte Realplayer scheint nicht alle Codecs zu unterstützen. Es liegt an Nokia, hier einen besseren Realplayer anzubieten. Ich liebäugele mit dem N900, das hat vielleicht aktuellere Mediensoftware.
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