Is it possible to play two audio files simultaneously.

Hi,
Is it possible to play two audio files simultaneously.
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.
Presenter is an add in for PowerPoint. The neat thing about it is that it provides for a Table of Contents type of functionality as you see is configured in Captivate. But the twist with that is you are able to do something you cannot do in Captivate. You are able to insert videos inside the TOC area!
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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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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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];
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CFStringRef fileString = (CFStringRef) [NSString stringWithFormat: fileName, self.recordingDirectory];
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Unless I'm mistaken, you need one AudioQueue object for each audio file you want to play. In one of my apps, I basically copied the code provided in the [AudioQueue programming guide|http://developer.apple.com/iphone/library/documentation/MusicAudio/Concep tual/AudioQueueProgrammingGuide/AQPlayback/chapter4_section_1.html#//appleref/doc/uid/TP40005343-CH3-SW1] and made a wrapper object for playback. So then you can do something like this:
MyAudioManager *manager1 = [[MyAudioManager alloc] initWithFile:@"file1.caf"];
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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 two FLV files simultaneously

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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.
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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);
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         // 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()){
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             audioFile = new File("junk.wav");
            // }//end if
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               targetDataLine.start();
               AudioSystem.write(
                     new AudioInputStream(targetDataLine),
                     fileType,
                     audioFile);
             }catch (Exception e){
               e.printStackTrace();
             }//end catch
           }//end run
         }//end inner class CaptureThread
}

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    } catch (IOException e) {
    } catch (LineUnavailableException e) {
    } catch (UnsupportedAudioFileException e) {
}

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                    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.

How do i play two sound file one after another

Hi All,
How do i play two sound file one after another using single AudioClip Component?.
Advance in thanks
Manivel

AudioClip gives you very little control over what happens to the sound, it also isn't capable of playing long clips or waiting until a clip ends. It will play multiple clips over top of each other.
To do what you want you should use the facilities of javax.sound. Here's a post with an example: http://forum.java.sun.com/thread.jspa?forumID=513&threadID=450768
There is also a tutorial, but its example fails for long clips.

Is it possible to play two audio files simultaneously.

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