Play cd audio labview
Bonjour,
J'ai trouvé sur le forum un VI permettant de lire les pistes d'un CD
audio avec LabView et ça fonctionne très bien (merci à son auteur).
Cependant, j'aimerais aussi avoir comment récupérer des informations
sur la lecture en cours, notamment la piste en cours de lecture, la
durée restante, le temps en cours, le nombre de pistes du CD inséré. Je
sais que que winmm.dll permet de récupérer ces informations, mais je ne
sais pas comment coder ça en LabView.
Je joins le fichier original qui permet la lecture des pistes.
Merci d'avance pour votre aide.
Hello,
I've found on this forum au VI which can play audio CD tracks with LabView and it works great (thanks to the author).
But I would to know how to recover the playing track, the elapsed time, the playing time, and the number of tracks on the cd. I Know that winmm.dll permit this functions but I don't know how to code that with LabView.
I join the original Vi which plays the audio tracks.
Thanks for help.
Attachments:
PlayCD.vi 22 KB
Hi,
It's useless to post 2 topics on the French and English forum for the same problem.
Somebody answers here, advicing to have a look on the web to find this kind of page
You need to use a call library node, double-click the node and then enter the correct parameters to call the DLL winmm.dll properly.
Browsing documentation will help you to find the function that fit your needs.
Cordialement / Regards, Matthieu B. @ NI France.
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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 */
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int printlevel = INFO;
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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;
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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;
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dis.readFully(temp, 0, HEADERSIZE);
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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;
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if (printlevel <= DEBUG) {
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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);
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mode = 2; // Ultra-wideband
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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
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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
} -
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Hi All, i want to play a audio file from the point of mark.
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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
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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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