WAV File - CD Burning Issue ?

Hey there !
I recently had a strange issue in which I am hoping that one here might be able to help me solve.
I just finished mixing a song ... saved it as a .WAV (PCM) ... went to burn it on a CD using Nero 8 Express and Nero gives me a error of a Possible Corruption?
I am sure that the files in the past have worked ?
I have resaved it and still no good ... am I missing something here ?
!! THANK YOU in ADVANCE !!

Steve is spot on in Audiomasters: 16 bit, 44.1 kHz, Stereo files only and probably no extra information. However, since you didn't specify, I'll also point out that you need to use the "Windows PCM" option, not one of the others on the Save menu.
Bob

Similar Messages

WAV File Burned to CD as Audio Plays SLOW

Hello!
I recorded a set of mine (I'm a DJ) using Stanton Final Scratch. The file is a WAV. I copied it to iTunes, then burned it as an audio CD. When I play the CD, it's slow, as in a 45 playing at 33.
I can play the file in iTunes, but as soon as I burn it, it's slow. Same thing if I convert the file to say an MP3. I can convert to Apple Lossless, but then if I burn that, same problem.
HELP!!!

Those are quite some encoder settings! I'm actually surprised you can play the files at all in iTunes.
If I had such files, I would probably resort to using something like Audacity to export the files to "normal" WAVs, using the settings I mentioned in my previous post (no floating point, etc.).
Anyway, we're on the right track, as the problem is in the properties of your source files.
EDIT: I presume you used Traktor FS to create the WAV files, so why not see if it lets you export the same files directly to more compliant WAV files you can use in iTunes?

Burning WAV files- HELP

New mac user here
I need help with this as soon as possible please, I have a deadline-
And this MAC is killing me, I am running into problems all over the place
Long story short
1- I sent a drum loop from garageband to itunes
2- I converted the AIFF drum loop audio file to WAV with Itunes
3- I burned the WAV file to a CD-R and tried to import the file to my mixer but it does not read it( it only reads WAV files)
I am stuck, my question is-
can itunes write to ISO9660 data disk?
How do I do this?
If not what are my other options, how do I do this?
Any step by step guide would be greatly appreciated

iTunes can burn three types of disks.
Audio CDs (to play in a traditional CD player)
MP3 CDs (to play in newer CD players which can also play MP3 tracks)
and
Data disks (CD or DVD)
An audio CD would certainly not be what you want, but a Data disk might. I suggest you use a CD-RW (which you can erase and re-try with if necessary).
Alternatively, there is another way. In stead of using iTunes to burn the disk do the following.
1. Insert a blank CD
2. drag the WAV file on to it
3. Burn the disk
This will burn an ISO standard CD with your WAV file on it.

Bounced (rendered wav files) file issues

Got a mix, all good, nice blend , nothing overpowering , good bass and kick balance and mids are just right( mix not mastered so i am sure mix could be improved) but ok for currenty purpose. Ok now bounce file(used power dithering option 1) , played back wav file now it sounds like sh.....t , bass is realy dominant kick is extremly loud and sound is cluttered. Why? should i not use the dither option? in other words after bouncing mix to wav the sound is no where close to the original mix even on same monitors used to reference mix originally, issues are more pronounce when playing back on other systems. Point me in right direction
thanks

First of all, if you don't have bit depth conversion dithering is absolutely unneeded.
Second, try using another audio format (AIFF or SDII, but I prefer the first).
Third, if you have a DAT recorder, try recording your audio card's analog stereo out directly onto the DAT then, if you need, re-import it digitally into Logic. (take care of DAT sampling rate, 48Khz, and of clock settings)
Fourth, some audio cards (like MOTU's one) can send a mixed stereo out back to the DAW software without cabling anything.
Let me know.
cheers
rob
Power Mac G4/533 PowerBook 12 Mac OS X (10.4.8)

Bounced WAV files's timestamp doesn't match timeline issue.

I'm mixing down some music for picture. In Logic Pro 8.01, the audio/video sync perfectly. But the bounced WAV files' timestamp does not match. For example, in Logic Pro/Video's timeline, it is starting at 01:00:00:00, but the resulting WAV file's timestamp is at 01:00:03:18. Or Logic's timeline start at 01:13:22:00, the resulting WAV file's timestamp is at 01:13:26:12. I checked it with Pro Tools and BWAVReader.
Logic is set at 29.97 Non Drop and 48kHz matching the DV file. Am I doing something wrong or is this a bug? Any help would be appreciated. Thanks

my preference would be to get Audition 3 (too bad we can't get an upgrade price from CS3 Production Premium to Audition 3...).
Are you sure. I got Audition 3 for US$99, IIRC, from CS2 Production Studio (contained Au 2.0). That was just after the release of Au 3, but not THAT long ago. With CS3 Production Premium, I would think that you would qualify even more than I did.
One point - I'm not using Reverb as an effect...
Then that was my error. I thought that you HAD added a bit of Reverb - sorry. And no, I was talking about the Effect, not the room ambience. Just a mistake on my part.
I am going to try a work around tonight... I am going to export each of the 2 audio tracks separately (with effects applied) and then bring those 2 WAVs back into PP3 to be mixed and just drop the original audio to see what happens
I often do similar to what you are doing - Upmixing. I know that the music industry is opposed to this with re-releases of music, where original multi-channel masters do not exist, but for my work, it is OK. I'll use duplicate Audio, just as you are, and then mix each 2-channel Clip a bit differently, using a touch of EQ, Reverb and Delay (most often), plus the spatial location of the sources for DD 5.1 SS mixing. Now, I am keeping the Effects to a minimum, but the Audio is room-filling, when I do it right, and with a light-hand. For my sources, I have found subtle is better, unless I need a panned SFX, or similar, and then I get to go a bit wild, like landing the helicopter on the back of the audience's heads!
Have you tried SoundBooth for the Audio mixing? I only used it in Beta, but had Audition, so never saw the use for it in my workflow. Who knew that Audition would soon be only a stand-alone?
Good luck, and please let me know what you find out with your tests. I can always learn more about the Audio-side of the street.
Hunt

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.

Why won't my .wav files show up in a 3rd party sound application after being exported as an omf file

Hi all, I've searched everywhere and can't find an answer, so hopefully someone out there will have mercy and help me!
I have Canon 7D footage (h.264 codec) and H4N Zoom audio (24bit 48khz .wav files) loaded into Premiere Pro CS 5. Editing on a Macbook Pro OS 10.6.8. I haven't added timecode.
My workflow, which I've done successfully several times in the past, is to sync the footage by creating individual sequences for each take (usually interview takes) and then bring in the audio and video and manually line/sync them up. I then link the two clips.
I then create a new sequence for my edit and then cut parts of the previously synced clips and then paste them into the sequence and edit away. After locking I export an omf (48khz, 24 bit, files as separate audio, aiff files, with 24frame handles). Then I import the OMF into Soundtrack Pro.
However this time, the majority of the sound clips do not show up in Soundtrack Pro timeline even though they are present in the omf folder as Aiff files. Normally you would think this was an issue with Soundtrack Pro, however one audio file did show up in Soundtrack Pro, but it was a sound file I had not linked to a video clip. Upon further investigation I found that that the audio clip that was not synced retained it's STE-014.wav name in the Premiere timeline. All the other audio I had synced to video had somehow taken on the video clip's MVI 0075 name.
I checked my old projects (the ones that I successfully exported to OMF) and even when I synced the audio and video, the audio retained it's STE.wav name. I don't know why the names changed; I did a test in my current project by cutting and pasting a synced clip at the end of the edit timeline and the audio retained it's STE.wav name. See pic.
After further tests, if the audio retains it's STE name it shows up in Soundtrack Pro (and Soundbooth as well), if it takes on the video MVI name it doesn't. I've tried exporting as an AAF and a Final Cut Pro XML file and it still doesn't work. I've tried unlinking and relinking the clips as well as making the audio offline and then relinking but still nada. I also cleaned the media cache to no avail.
If I render and replace the audio in the edit timeline, the name doesn't change, but the audio file does show up in Soundtrack Pro and Soundbooth, but unfortunately I need handles to edit the sound so this doesn't help me much.
Any ideas? and thanks!

Yes, it has always worked in the past. The past projects audio clips do not change names. I obviously did something different in this current project, but unfortunately I have no idea what.

Itunes won't import or convert .wav files?

I have a bunch of .wav files that I wanted to import to itunes, so that I could convert/burn them to cd's. I don't have any problem playing the files in Quicktime. But when I import them, although it appears to work fine, if I try to play it, it won't play, it skips to the next song. When I try to convert to mp3, it gives me the message:
Error occurred while converting the file "xxx".
The disk could not be read from or written to.
The files are just on my desktop, not on a CD. I don't have a problem converting other files. What could be causing this?

Same complaint here. I use an 80gb iPod Classic with an Alesis iMultimix USB8 live mixer to record performances at our local open mic. Starting in April, the WAV files from the iPod would automatically stream into iTunes when I connected the iPod, and it was a snap to select a whole evening's worth and convert the lot to MP3 to save archiving space (by a ton). Starting in September, I seem to have some kind of random corruption of these files.
Some hang up the program immediately upon being selected and the "convert to mp3" chosen from the Advanced menu. Others start to convert and then hang up, usually with 10-15 seconds left to convert. Others convert just as before without problems. I have had to force-quit iTunes a number of times to unlock the system. This is getting really old, and what was a semi-automated process has become one of picking through the files one by one, force-quitting when it hangs, then deleting the culprit when I re-open iTunes. I'm losing some irreplaceable recordings, which pretty much negates the whole purpose of capturing everything every week.
I don't recall when I made the last prompted update of iTunes, but it very likely could have been just before this started happening.
Another glitch in this recording method that's a bit off topic: there really should be a two-step procedure to delete a voice memo. More than once I have deleted an abortive performance, then recorded the next one. When I pause it and the menu comes up, it's easy not to notice that the selected option is still "delete". I make about 20 of these recordings every week, and it's my habit to select "stop and save" quickly so as to get the next one started. If we got a second screen asking if we're sure we want to delete, it would be easy to catch.
The voice memo feature is great, especially with a mixer than can use the iPod as a hard drive to capture live performances, but the software needs work!

.WAV files no longer readable after Yosemite update

The issue:
I recently did the switch to Yosemite. I now find some .WAV files to be no longer readable through FCP X, Preview app, QT, etc. BUT, I can open them through VLC, and also Soundtrack Pro. The problem:
An FCP X project with this .WAV files has already been edited, with multiple cuts (46 to be exact) throughout the 35 minute timeline. The file is still visible on timeline with all cuts but is now silent. Looking to relink that audio with a working file. The original files' timecode starts at 09:46:30:09, with 7 channels of audio. What I tried to do:
Open in Soundtrack Pro and simply "Save as" a new .wav file. Still not readable. Then brought the original file in to a new Multitrack Project in Soundtrack Pro, exported to .wav, which is now readable by all programs, but cant be relinked since info are not matched (channel count and timecode) "Incompatible file.
The original file and new file have different audio source and channel counts.
Relinked files must have the same media type and similar audio channels as the original files, and must be long enough to cover all the clips that reference the files."
My system:
FCPX (10.1.3)
OS X 10.10 (Yosemite) iMac (27 inch, Late 2012)
3.4 GHz Intel Core i7
32 GB 1600 MHz DDR3
NVIDIA GeForce GTX 680MX 2048 MB

Thanks Martin. Thats just what I'm going to do. For the record, I have been dealing with Apple support concerning this issue for the last month. It's been a frustrating and exhaustive process and the outcomes has been underwhelming to say the least:
It seems that under OSX Yosemite, Apple Core Audio , which plays audio in all the apple apps: QT, iTunes, FCPX, Logic Pro X etc, can no longer read multi-track WAV files larger than 2GBs. I tested this extensively myself, and the support staff member I was dealing with was able to re-create the problem at his end.
Apps that do not use Core Audio CAN play the WAV files: QT7, Adobe Premiere, Audacity etc.
Using this information the support member helping me escalated the issue to the Apple developers teams for both FCPX and OSX. He said he was really unimpressed by their response, and that while they acknowledged that there 'may' be an issue with Core Audio they didn't care enough to investigate it. He said that while a fix may come in the future there is no timeline for it to happen. He said he wasn't allowed to recommend that I downgrade, but he then rather embarrassingly admitted that I have no choice if I want to have a functioning project again.
Which makes a mockery of my decision to use FCP over Premiere based on the assumption that Apple would better support their software. FCP has pretty much ****** me off right from the start with its babyish interface and glitchy behaviour. Now I have to downgrade my system to Mavericks, in order to limp to the finishing line with this edit, before I will be ridding myself of this junk for good.

"This wav file could not be read"

I have a DL project with 30 timelines. It has been playing and burning fine for the last 4 months--I've just been tweaking menus and replacing some of the assets with newer encodes.
Yesterday when I attempted to preview I got the message "this wav file could not be read." There are 35 .wav files in the project. This morning I did a "save as" on the project and systematically removed the wav files from the timelines two at a time until I found what was causing the problem (naturally it was the very last one). This was not a wav file that had been replaced in recent months, it has been in the project from the beginning
Then I opened the original project and deleted only that .wav file from the timeline and the project previewed fine. So I deleted the .wav file from the project and then imported the same .wav file again, placed it in the timeline and the project works fine.
Can someone tell me what would cause this behavior? Is there some sort of corruption in the asset and I'm going to have the same problem again? Or is it more likely some corruption in the project, and if so what could it be or what should I do about it?

I have just replaced all WAV files in the project with ac3 files("reaplace asset" command).
Still, when building DVD, god damned Encore CS4 mumbles "this WAV file could not be read."
WTF?
I looked through several topics on the Adobe forums where people ask the same question. There is no one a single answer! So, I'm back to Sonic Scenarist!
No time for "dancing with a tambourine," around a blunt Adobe.
And by the way, when Encore telling "This wav file could not be read" -- WHY it's not telling me WHICH ONE file? Do I have to guess or ask a fortune-teller?

Cannot hear imported wav files in the output

Hi, I am using Macromedia Captivate. Version 1.01.1447
I have imported a wav file and when I am playing the slide containing the audio file everything is fine, in particular I hear my audio file. However, when going to the preview of the entire slideshow (or just 5 slides), or if I preview the show in the browser, I do not hear anything. Also, when exporting the movie as a standalone file, I don't hear anything.
Does anyone know a solution for this issue?
Thanks!

If you can get to the instrument editor, drag and drop your regions or files onto the keyboard and answer the following questions. To preview it have an EXS24 loaded up with EXS Instrument #5678 (or whichever number it happens to be.

ITunes "organized" my wav files.

I have approximately 2000 wav files on an external hard drive. All the songs were imported using iTunes. Because of hardware issues, the internal HD was reformatted and the OS reinstalled. When I "add(ed them) to library" without "keep iTunes Music folder organized" checked, the songs appeared in the iTunes browser without the artist or album listed. So I checked the "keep iTunes Music folder organized." Well of course iTunes placed all the wav files in a single folder. It appears all album information and artist information was lost. Does anyone have any idea about how I might reorganize the files? Thank You.
This is an obvious flaw in iTunes. Most likely easily correctable. I'm not sure why apple has not taken the time to fix this bug.
PowerPook G4 Mac OS X (10.4.5)

Sorry - but seems to be a hard work to come.
What about the nameing of the wav files? Did you originally select "Create file names with track number"?
The problem in you case: while all track information are stored within mp3 + aac files as tags, those information will be store externally and linked to wav + aiff files: reformatting your hd erased that information.

How do I stop iTunes from automatically opening .wav files

I recently arranged to have voicemails to a business number sent to me in the form of .wav files that I check at the end of the day from my home PC. I was disappointed to learn that when I open the email and double click the .wav file it automatically imports it to my music library in iTunes (also selecting to be synched). I am just trying to get these files to open in Quicktime. Any help?

Can you tell me how? I am having the same issue, but I think mine is opening automatically whenever I download any new music files.

Problem in playing back mp3 which was recorded as wav file

Hi all,
I have recorded sound using my program. It uses classes like AudioSystem. And it's
able to record and play also. but in my site I have an mp3 player. So i thought of converting the recorded wav file into mp3. For this i used LAME utility in Linux. And it was converted. Also it is being played in normal players like Winamp. But when i try to play this converted mp3 file using my player in my site it's having problem. Why is it like that?
Should i do something while capturing the sound from microphone so that it can be converted into mp3 format??
Or
Can I directly capture voice to an mp3 file??
In my program,
targetDataLine.open(audioFormat) where audioFormat is like
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
Should I do something more or change something??
Please help me...
Thanx and Regards,
sand...

Thanks Walter.
Unfortunately, the same issue happens... In whichever window I select "Bounce" from -- Environment, Audio Mixer, or even FILE > BOUNCE -- a window titled "Bounce 'Output 1-2' " pops up. I'm guessing this is what you mean by the "Bounce Output Window." But, there I'm only able to name the file in the "Save As" field, choose where to bounce the file to on my harddrive, and then click on "Bounce" in the bottom-right of the window. I have no tabs or anything else to choose from. So I'm still a little stumped. Any additional thoughts?
I really appreciate the help.

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++;
    }

WAV File - CD Burning Issue ?

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