Playing a .wav file from email attachment

I am trying to play a .wav file that is attached to an email, and i get "The document conversion failed". Am I missing something? What do I need to do to be able to play .wav files that are sent to me via email?

I have the same request.
We have a new Panasonic phone system that sends out .wav voicemail files.
Our company's Curve's play the wav fine.
Our company's BOLDs ... DO NOT.

Similar Messages

Subsequent play of WAV files from HTTP

Hi,
I have to subsequently play many WAV files from HTTP in a nokia series 60 emulator.
Everything is ok for the first two files, but at the third I catch an exception:
"java.io.IOException: exceeded the configured maximum number of connections"
I tried two ways to load the files from HTTP:
Manager.CreatePlayer(InputStream...)
Manager.CreatePlayer("http:\\...")
Below there is the code of the two methods I use. Please note that I try in every way to close the connection before open a new one!!!
public void Method_1(String src, String type){
int iLen=0;
is = null;
httpCon = null;
byte[] bBuf;
try {
if (p != null) {
if (p.getState() == Player.STARTED) {
p.stop();
p.removePlayerListener(this);
p.deallocate();
p.close();
p=null;
if (is!=null){
is.close();
is=null;
if (httpCon!=null){
httpCon.close();
httpCon=null;
httpCon = (HttpConnection) Connector.open(src);
if (httpCon.getResponseCode() != HttpConnection.HTTP_OK) {
System.out.println("Bad response code");
return;
System.out.println("Connection OK! (audio)");
iLen = (int) httpCon.getLength();
System.out.println("audio file length : " + iLen);
is = httpCon.openInputStream();
p = Manager.createPlayer(is, type);
p.addPlayerListener(this);
p.setLoopCount(1);
p.prefetch();
p.start();
vc = (VolumeControl)p.getControl("VolumeControl");
catch (MediaException ex) {
ex.printStackTrace();
catch (IOException ex) {
ex.printStackTrace();
public void Method_2(String src){
try {
if (p!=null){
if (p.getState() == Player.STARTED)
p.stop();
p.deallocate();
p.close();
p=null;
p = Manager.createPlayer(src);
p.addPlayerListener(this);
p.setLoopCount(1);
p.start();
vc = (VolumeControl)p.getControl("VolumeControl");
catch (java.io.IOException e) {
e.printStackTrace();
catch (MediaException e) {
e.printStackTrace();

My question is:
how can I load (from a HTTP server) and play more than two WAV files without crashing?
Do I have to close the connection in a better way in order to re-open it without problems (and more than two times)?
It is simply an emulator bug?
Thanks.

How do I play the .wav file google mail sends me on my Iphone 5?

I have cable phone service that sends out a .wav file of any voicemails I recieve to my e-mail account. I use google mail on my Iphone, and cannot play the .wav file that is attached to that e-mail. When I click on the "audio message box", it brings up a circle containing a south-west facing arrow which has a slash through it.

I have the latest software (6.1.4) and I can play the attachment if I go through "Mail" instead of "GMail". I am still not certain why I have a widget called "Mail".
Inside the "Mail" app, there are it's own Inbox, VIP, Flagged, etc which are all in color. Further down the list is my GMail boxes and sub-boxes. They are all grayed out, but still funtional.
Do I have something set up wrong causing me to be unable to play the .wav files that come to me through gmail?

How can i dowload pdf,docx attachment files from email app?

There is no option to download pdf,docx files from email app.

Hi francis29,
Viewing and downloading unsupported file-types is not possible in the current version of Firefox OS. We are aware of this issue and our engineering team is working on adding this feature.
You may look at [https://bugzilla.mozilla.org/show_bug.cgi?id=825318 bug 825318] to track the development of this feature.
- Ralph

Need Help playing a wav file on PDA

Does anyone have a SIMPLE example of playing a wav file on a PDA using LabView PDA. The only working example I can find from NI "Advance Sound Example for LabVIEW PDA for Pocket PC" is a jumbled of confusing functionality. I'm mainly looking for a simple "play" example, but a simple "record" example would be helpful also.
Thanks - Bruce

Hello,
Well, I looked through the example code I tried to break it down to the point where you could create a simple Play VI. Unfortunately, I don't have the PDA target, so I'm uncertain if the little example that I have attached will run or not. If not, I also took a screen shot so you can see what VIs I used. I'm pretty sure I only used VI's that I found on the example itself.
Good luck.
Attachments:
Play_Wav_PDA.vi ‏55 KB
Play_Wav_PDA.bmp ‏1029 KB

Broadcast Wav files from PT to Logic not moving to original record position

Okay, my system needs upgrading but I'm stable (at least I think I am til now!) I have a TV network sending me broadcast wav files from PT. They shoot their video on a 24 hour clock. I'm to be editing and mixing (in Logic) their audio tracks from PT and send them back as broadcast wav files for them to sync back to their video. After years and years of recording and producing music, this TV world is a first for me. 2 issues at hand that are causing trouble.
NUMBER 1: The wav files will not snap to original record position. I keep getting an error message about "out of range" and of all craziness with a negative SMPTE number. ??? The prompt suggests an offset error. But there isn't I can find. Start time is set to 0:0:0.0.0 here and at the TV studio. They film at 29.97d. That matches as well. Sample Rate is correct at 48k 24bit. I've uncovered every thing I can think of. I did have to set the tempo down to 5 BPM in order to see a full 24 hours in SMPTE.
NUMBER 2: I can't figure out how to see the timestamp start time of the audio files in the audio window. I have every option turned on in the VIEW option, I can see length, but I see nothing about start time of original record time. Am I missing something simple here? Seems like a very standard option that should be easily accessible. It would allow me to prepare the location on the time line.
Overall, I'm hoping I can do all the above with my current rig but realize that the clock is ticking toward a whole new system with Logic 9.
Help please. Don't want to lose this client.

Sure do appreciate everyone's willingness to help. Setting SMPTE start to 24:00:00.0.0 didn't work either. I confirmed with the studio manager at the TV station also that they run a 24hour clock with 0 being midnight.
I'm really hoping that the error is on their end.
*It would help me if anyone knew how to VIEW the Timestamp info attached to the wav files. That would at least give me an approximate idea of where they should be.*
I'm supposed to be editing and mixing the voice dialog from numerous different scene takes that were shot throughout a day of filming. It's critical that I can see all these different takes that were captured over hours of open recording. Some of the files are almost 4 hours long. It's a big project and I'm sure hoping I can stay inside Logic to get it all done with excellence.

Preview 6.0, no send file as email attachment?

Does anyone know where they put the send file command? I need to email pdfs from Preview all the time.
There used to be a "send file as email attachment" command under the edit menu. It's gone. Anyone have any suggestions?

Handsomfreddy wrote:
Sorry, I dont understand what you are referring to. Can you calify please "undo the changes."
Thanks,
Andy
Didn't you just post this (below)
Handsomfreddy wrote:
Very bizarrely Parallels had changed this Mail Preference to some weird Windows program [Disk Clean up or something equally random] thus preventing me from emailing anything from any program.
That's the change I am referring to.

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 a wav file (byte array) using JMF

Hi,
I want to play a wav file in form of a byte array using JMF. I have 2 classes, MyDataSource and MyPullBufferStream. MyDataSource class is inherited from PullStreamDataSource, and MyPullBufferStream is derived from PullBufferStream. When I run the following piece of code, I got an error saying "EXCEPTION_ACCESS_VIOLATION (0xc0000005) at pc=0x7c9108b2, pid=3800, tid=1111". Any idea what might be the problem? Thanks.
File file = new File(filename);
byte[] data = FileUtils.readFileToByteArray(file);
MyDataSource ds = new MyDataSource(data);
ds.connect();
try
    player = Manager.createPlayer(ds);
catch (NoPlayerException e)
    e.printStackTrace();
if (player != null)
     this.filename = filename;
     JMFrame jmframe = new JMFrame(player, filename);
    desktop.add(jmframe);
import java.io.IOException;
import javax.media.Time;
import javax.media.protocol.PullBufferDataSource;
import javax.media.protocol.PullBufferStream;
public class MyDataSource extends PullBufferDataSource
    protected Object[] controls = new Object[0];
    protected boolean started = false;
    protected String contentType = "raw";
    protected boolean connected = false;
    protected Time duration = DURATION_UNKNOWN;
    protected PullBufferStream[] streams = null;
    protected PullBufferStream stream = null;
    protected final byte[] data;
    public MyDataSource(final byte[] data)
        this.data = data;
    public String getContentType()
        if (!connected)
            System.err.println("Error: DataSource not connected");
            return null;
        return contentType;
    public void connect() throws IOException
        if (connected)
            return;
        stream = new MyPullBufferStream(data);
        streams = new MyPullBufferStream[1];
        streams[0] = this.stream;
        connected = true;
    public void disconnect()
        try
            if (started)
                stop();
        catch (IOException e)
        connected = false;
    public void start() throws IOException
        // we need to throw error if connect() has not been called
        if (!connected)
            throw new java.lang.Error(
                    "DataSource must be connected before it can be started");
        if (started)
            return;
        started = true;
    public void stop() throws IOException
        if (!connected || !started)
            return;
        started = false;
    public Object[] getControls()
        return controls;
    public Object getControl(String controlType)
        try
            Class cls = Class.forName(controlType);
            Object cs[] = getControls();
            for (int i = 0; i < cs.length; i++)
                if (cls.isInstance(cs))
return cs[i];
return null;
catch (Exception e)
// no such controlType or such control
return null;
public Time getDuration()
return duration;
public PullBufferStream[] getStreams()
if (streams == null)
streams = new MyPullBufferStream[1];
stream = streams[0] = new MyPullBufferStream(data);
return streams;
import java.io.ByteArrayInputStream;
import java.io.IOException;
import javax.media.Buffer;
import javax.media.Control;
import javax.media.Format;
import javax.media.format.AudioFormat;
import javax.media.protocol.ContentDescriptor;
import javax.media.protocol.PullBufferStream;
public class MyPullBufferStream implements PullBufferStream
private static final int BLOCK_SIZE = 500;
protected final ContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW);
protected AudioFormat audioFormat = new AudioFormat(AudioFormat.GSM_MS, 8000.0, 8, 1,
Format.NOT_SPECIFIED, AudioFormat.SIGNED, 8, Format.NOT_SPECIFIED,
Format.byteArray);
private int seqNo = 0;
private final byte[] data;
private final ByteArrayInputStream bais;
protected Control[] controls = new Control[0];
public MyPullBufferStream(final byte[] data)
this.data = data;
bais = new ByteArrayInputStream(data);
public Format getFormat()
return audioFormat;
public void read(Buffer buffer) throws IOException
synchronized (this)
Object outdata = buffer.getData();
if (outdata == null || !(outdata.getClass() == Format.byteArray)
|| ((byte[]) outdata).length < BLOCK_SIZE)
outdata = new byte[BLOCK_SIZE];
buffer.setData(outdata);
byte[] data = (byte[])buffer.getData();
int bytes = bais.read(data);
buffer.setData(data);
buffer.setFormat(audioFormat);
buffer.setTimeStamp(System.currentTimeMillis());
buffer.setSequenceNumber(seqNo);
buffer.setLength(BLOCK_SIZE);
buffer.setFlags(0);
buffer.setHeader(null);
seqNo++;
public boolean willReadBlock()
return bais.available() > 0;
public boolean endOfStream()
return willReadBlock();
public ContentDescriptor getContentDescriptor()
return cd;
public long getContentLength()
return (long)data.length;
public Object getControl(String controlType)
try
Class cls = Class.forName(controlType);
Object cs[] = getControls();
for (int i = 0; i < cs.length; i++)
if (cls.isInstance(cs[i]))
return cs[i];
return null;
catch (Exception e)
// no such controlType or such control
return null;
public Object[] getControls()
return controls;

Here's some additional information. After making the following changes to MyPullBufferStream class, I can play a wav file with gsm-ms encoding with one issue: the wav file is played many times faster.
protected AudioFormat audioFormat = new AudioFormat(AudioFormat.GSM, 8000.0, 8, 1,
            Format.NOT_SPECIFIED, AudioFormat.SIGNED, 8, Format.NOT_SPECIFIED,
            Format.byteArray);
// put the entire byte array into the buffer in one shot instead of
// giving a portion of it multiple times
public void read(Buffer buffer) throws IOException
        synchronized (this)
            Object outdata = buffer.getData();
            if (outdata == null || !(outdata.getClass() == Format.byteArray)
                    || ((byte[]) outdata).length < BLOCK_SIZE)
                outdata = new byte[BLOCK_SIZE];
                buffer.setData(outdata);
            buffer.setLength(this.data.length);
            buffer.setOffset(0);
            buffer.setFormat(audioFormat);
            buffer.setData(this.data);
            seqNo++;
    }

How to play a wave file through the DAQ board?

Hi there,
we want to play a wave file that is saved on the harddrive via the DAQ board. Means we want to connect a amp to the analog output of the DAQ to amplify the sound and play it through a speaker.
We couldn't find any VI that does that. Can anybody tell us how to play a wave file, or perhaps even send us a VI that can do that?
Thanks a lot for your help
Martin - Physical Optics Corporation

Hi Martin,
The only example that I could find on how to do this was using our
older Traditional DAQ drivers version 6.9.3. If you're using Traditional
DAQ drivers, the vi can be found here.
The example reads a 8-bit mono .wav file (an example .wav file is included in the .zip file) and writes it to the analog out of your DAQ card. I've updated the
example to work with our newer DAQmx drivers and attached it here.
I hope this helps,
Paul C.
Message Edited by Paul C. on 08-14-2007 05:47 PM
Attachments:
daqmx_wav_file_to_ao_output.zip ‏27 KB

Play a .wav file fractional

hi,
i´m looking for a solution to play a .wav file.
the special what i need is, that i want to jump to concrete points on the timeline.
or can i seperate the timeline of a .wav file?
martin

Hi Martin
this could help you starting. For the case you can't open LV 8.2- VIs, i attach the BD- picture.
greets, Dave
Message Edited by daveTW on 06-14-2007 10:26 AM
Greets, Dave
Attachments:
play sound part_FP.png ‏10 KB
play sound part.vi ‏50 KB

Send file as email attachment not working

any help would be greatly appreciated...
When I click "send file as email attachment" it will open outlook with the PDF attached but when I click send on the email, the email closes like it is sent but never actually gets sent (nothing in sent items and recipient never gets the email)
Adobe Acrobat 11 Pro (Creative Cloud Suite installed)
Outlook 2013

Hi sandyw26405588,
Could you please try sending PDF as email attachment via another email apart from Outlook.
Are you able to send the file successfully?
Also, go under 'Help> Check for Updates' and ensure if you are using the latest version of Acrobat XI i.e. 11.0.10
Have you tried sending any other PDF via Outlook?
Let me know.
Regards,
Anubha

How do i make it so iTunes plays the music files from where i have chosen to store them?

how do i make it so iTunes plays the music files from where i have chosen to store them as opposed to it scattering copies of my files into a myriad different folders based on the albums they came from when i'm not even interested in keeping whole albums?
I recently converted myself to mac after having been a windows formatted person since the beginning of time. I use last.fm and the client scrobbler only supports iTunes on the mac (scobbling since Nov 06 and i'm now heavily dependant upon it as a means of discovering new music) - if you don't know what that means then take it for granted i have no option other than to use iTunes as my media player whn using the mac. And almost immediately upon using iTunes i was filled with horror.
My tale of woe as follows: I copy all my mp3 files to folder X on my new mac. I copy my simple playlist in to that same folder X. The simple playlist simply lists all the files in folder X in the order i want to hear them in... simple right? And what happens the moment i start playing my simple list? Well you probably all ready know; itunes throws copies of all those files in to the air, scatters them to the four winds so each of them fall into a little folder of their own representing the album they might once have come from. I don't keep whole albums. For me the 'album experience' is a ruse to trick one into listening to weaker material. I only keep the concentrated goodness. I only put up with albums in a previous lifetime because in the days of vinyl i had no other choice apart from spending hours compiling mix tapes. Now i am liberated from these restrictions and have been for the 15 or so years i've been using a computer to listen to music. Until along comes iTunes forcing me backwards in time, frog marching me back into the past.
That simple list i mentioned is my one and only playlist. This single playlist has been evolving for 15 years as tracks are added and removed and their position in the list changes because that song sounds amazing when heard after that other song whereas hearing this song after that song would be like pouring custard over your sausages (for me shuffling is an abomination). And so here is how i've managed this list and managed synchronization of said list and listed files with my mp3 player (currently a Zen).
Adding a new file: After deciding that song is a keeper i will copy it into that simple folder that holds all my music and drop it in to a position on the list i deem most suited for it. Incidentally over this entire 15 year (i'm pretty sure it's been around that long) my media player has been winamp but for the reason previously mentioned i am unable to use winamp in mac world.
Removing a file: fortunately winamp has an option to simultaneously remove a file from the list and the hard drive. And naturally so because after all if i no longer like a song then i want it expunged completely from life or at least from my hard drive and playlist. In the brave new world of iTunes i have to remove from the list there then delete the file from where iTunes copied it from and then the place iTunes copied it to.
Editing the list: drag an item in the list to it's preferred location and select to save it as i would for any other file using any other application... well desktop type application as opposed to one that uses a database as its primary means of storage.
Syncing with the mp3 player: copy drag drop playlist file from the simple folder on the hard drive to the simple folder on the mp3 player as i would for any other file. And then also copy drag drop the most recently created files from simple folder on hard drive to simple folder on player after having ordered them by creation date so as to make it easier to select the most recently created. After a while the mp3 player will fill up with the files no longer on the list so i will delete all files from player then re-copy all flies from simple folder on hard drive to simple folder on mp3 player.
I do not make changes to the list on the mp3 player so synchronization needs only to go one way: from laptop to player.
So all very simple. Apart from editing the positions of files in the list everything is managed by copy drag dropping and deleting files using windows explorer.
But i'm in a mac and circumstances have forced me in to the corner of using iTunes. And this whole iTunes situation is terrifying for me as over the years i've heard horror stories about it deleting all of someone's music files. Perhaps this may have been by using the software incorrectly but even so, the very fact that it appears possible for this to happen simply by using the software incorrectly and the fact that it happens with some degree of frequency given the number of reports of this happening that i have heard of over the years on music discussion forums fills me with dread that i have to now use this software. And given the software's blatant disregard for the user's choice as to where to store his/her files it's no surprise that this happens. The software is pointlessly doing dangerous things with my data. Please make it stop.
Please can i just at the very least make iTunes play the music from the folder i have chosen to put my files in and not copy those files here there and everywhere without my say so? Thankyou.

option discovered in preferences. Really this and any feature that moves deletes or edits a users data should as far as possible be OFF BY DEFAULT. It shouldn't by default and without the users specific say do this dangerous and unnecessary thing without even letting the user know what it's doing!!! And then cause me a few hours (as i'm new to mac) searching for the option so as to switch it off. If i hadn't noticed the status cage declaring that it was copying files then i might never have twigged that this insane thing was occurring. And if i hadn't of noticed i would have been left maintaining the contents of the folder i copied the files to, the folder which as far as i was informed and so believed was also the location of the music files i was playing. How does Apple justify this 'genius' piece of software non-design? Surely it will hurt no one if this was off by default for new users - overall at least i dare say it would cause a lot less distress in the future for the unsuspecting public at large. Thanks for the support.

How to delete files from an attached camera after downloading?

New to Aperture 2. I am sure to have a lot of exploring to do but for now I am looking for a way to delete files from an attached camera after downloading like you can with iPhoto. Is this possible?

While I might be a bit paranoid, I prefer to format in the camera after verifying that the import came out OK. It is a small price to pay to make sure that my images are actually on the computer. I don't do anything fancy, just make sure that the thumbnails come up correctly, and maybe perform an edit.
While it was not my images, thank goodness, I did get bit in a similar fashion. I use a "watch fob" style GPS unit for Geocoding. I have hoodahgeo configured to delete my tracks after import. Well, after one session I forgot to update my images with the gps information. As you can guess, when I re-opened HoodahGeo to try and re-geotag, the tracks were gone, so I was out of luck.

I can no longer play quicktime mov files from a canon firestore

I can no longer play quicktime mov files from a canon firestore from an HDV camera that play on other macs and play in vlc. It happened all of a sudden. Files that I could play before no longer play. It seems to be something wrong with the quicktime codec management because the files do not play in premiere and final cut pro. In FCP it says: File Error: 1 File(s) recognized, 0 access denied, 1 unknown. These files work on other computers. I have four drives and it doesnt matter which file i attempt to open on each drive, these files are unreadable on this system. Please help. Possible issues might be a corrupted index, but i dont understand how that can go across four different drives.
Model Name:          Mac Pro
Model Identifier:          MacPro4,1
Processor Name:          Quad-Core Intel Xeon
Processor Speed:          2.66 GHz
Number Of Processors:          1
Total Number Of Cores:          4
L2 Cache (per core):          256 KB
L3 Cache:          8 MB
Memory:          8 GB
4 SATA drives

Apple kindly updated Quicktime and rendered Firestore files unplayable. I think a recent update of iTunes was the culprit.
Solutions: What I did. Wipe the system disc. Re-install Snow Leopard. Re-install FCP7. Don't update Quicktime or iTunes to anything after Jun 2011. QT v7.6.6 is the last good version. That'll get your files back.
A colleague tried a different approach. For this you need a pre-July 2011 version of the file QuickTimeComponents.component. It is in the system/library/QuickTime folder. Delete your current one and drop in an old one.
However if you're working with Final Cut X, I think you screwed until Focus and Apple work something out. My address is on my contacts page at teevideousadotcom if you need any further help.

Playing a .wav file from email attachment

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