Is it possible to play an audio file out of labview (using myDAQ)?

Hi all,
I am using a myDAQ (first time). I am trying to play an audio file at a certain time in my VI.
All the examples I have seen use the audio in and then play out of the audio out port. Is it not possible to simply find the file location and then play it out of the audio out under the given conditions in the VI or even import the sound? I am not trying to play large files or large quantities such as songs. I only want a congratulatory sound when the task is complete (2/3s).
Any help would be is appreciated (even to say it isn't possible!)
Cheers
Solved!
Go to Solution.

Why no just play a WAV file? Use Play Sound File.vi as shown here.
Richard

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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!
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Captivate Wish Form/Bug Reporting Form
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Is it possible to edit/change audio files in a published html5 file?

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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 !!!
 AudioPlayer.player.start(as);
 return as;
Can you give me any pointers to how i code that fast forward bit.
many thanks
David
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
}

Playing the audio file from the point of mark

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
Thanks,
Ron

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Hi Everyone...
I would like to play an audio file (edc.caf) that is located in this directory....
/DCIM/TONY/IOW/AUDIO
When i run it through the NSURL processor....
[code]
NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath];
[code]
the resulting value of soundFileURL is
file://localhost/DCIM/TONY/IOW/AUDIO/edc.caf
But i do not hear any audio out of the iPod touch.
I've tried two different methods of playing the audio,
and both are failing.
I know the audio is being recorded, as i can bring it back to the PC and play them fine.
My initial thoughts are that i cannot find location of the file, but i am not sure.
Can anyone confirm the location is correct???
thanks
tony

Small edit....
the resulting value is a little different than what i posted....
the actual string is...
file://localhost/var/mobile/Media/DCIM/TONY/AUDIO/edc.caf
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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.
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Hi
I've done so, and the method gets called without any problems, but doesnt play anything.
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Ah',..okay. I constructed the DataSource like follows:
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I make the modification like you suggested. (into the DataSource, more exactly for method getStreams()).
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Thanks, that works, but it's not exactly what I'm looking for. Rather than have to search for the file in Finder and then hit the space-bar, I'd like to be able to simply double-click on an audio file, let it play in iTunes, but not be copied to the iTunes folder and remain in the iTunes browser until I delete it.
Also, I need to find a way to keep voice memos from residing in iTunes. I use them a lot, and they just clutter it up.
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I want to play video on my computer to make some analysis to frames,the problem that I face ,I can't change video frame rate using labview,but I can change frame rate to the video out of labview using some program

HI All
I want to play video on my computer to make some analysis to it's frames,the problem that I face ,I can't change video frame rate using labview,but I can change frame rate to the video out of labview using some program .
I used IMAQ AVI Read Frame VI
for example I have avi video It's frame rate is 25 fbs ,my image processing code is very fast that can process more 25 fbs,so I want to accelerate video acquisition

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I looked into this further, and reading an AVI file into LabVIEW faster than its frames per second won't be possible. LabVIEW could read in frames faster than 25fps, but because it will be pulling the available frame at that point in time this would just give you duplicate frames. If you want to be able to read in frames at faster than 25fps, you would need to speed up your AVI file before reading into LabVIEW.
There's a good shipping example to show how to read in from an AVI file, "Read AVI File.vi". You'll notice that they add timing to make sure that the while loop runs at the right speed to match up with the frames per second of the file being read. This is to make sure you're not reading duplicate frames.
Thank you,
Emily C
Applications Engineer
National Instruments

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Hi,
I am not familiar with audio files. I guess MP3 file has extension .mp3
can i still play the AudioClips?
Thanks!

It's got to be. It not built into standard java, but you can probably find classes for it on the web.

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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.
*NOTE: I have only tried this in the simulator, but I'm assuming it will do this on the real thing so I haven't bothered to try.

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:
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MyAudioManager *manager2 = [[MyAudioManager alloc] initWithFile:@"file2.caf"];
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I am not able to play it using QuickTim and VLC Player.
Can someone please help me how to play this?
Regards
Sharath Arun

Is that file played in a tab in the current window or in a separate pop-up window?
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*http://kb.mozillazine.org/JavaScript#Advanced_JavaScript_settings
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Is it possible to play an audio file out of labview (using myDAQ)?

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