WAV File Soundkits into GB.,...

I just gotta mac....and i switched over from pc...and i have a bunch of cd sounkits wit files in WAV format....i bought them on ebay...and the ebay seller says they would work in garage band.....how do i go about doin this???
heres a link to te item: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=190049668820&ssPageName=MERCVIC_RSCC_Pr4_PcY_BIN_IT&refitem=190045560489&itemcount=4&refwidgetloc=closed_vie w_item&usedrule1=StoreCatToStoreCat&refwidgettype=cross_promotwidget

If they are in WAV format you'll need to convert them with Apple's Software Developer's Kit. You can find a link to Apple's site which includes the Apple Loops SDK here:
http://www.thehangtime.com/gb/gbfaq2.html#soundtracklooputility

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Loading wave file data into array

Greetings
For starters, I'm still learning java, so forgive me if this is a trivial question. I've been using the documentation on the java sound api as a guide. What I am trying to do is to load a wav file data into a byte array and eventually I will need it to have it in an int array so I can pass the data into a demodulation algorithm to display as an image. This wave file is a recorded signal from a NOAA satellite. Here is my code thus far:
public boolean loadFile(File soundFile) {
          int totalFramesRead = 0;
          boolean result=false;
          File fileIn = new File(soundFile.getPath());
          try {
               AudioInputStream ais = AudioSystem.getAudioInputStream(soundFile);
               int bytesPerFrame = ais.getFormat().getFrameSize();
               // Set an arbitrary buffer size of frames.
               int numBytes = 1024*bytesPerFrame;
               byte[] audioByte= new byte[numBytes];
               try {
                    int numBytesRead = 0;
                    int numFramesRead = 0;
                    // Try to read numBytes bytes from the file.
                    while ((numBytesRead = ais.read(audioByte)) != -1) {
                         // Calculate the number of frames actually read.
                         numFramesRead = numBytesRead / bytesPerFrame;
                         totalFramesRead += numFramesRead;
                         return(result= true);
               } catch (Exception ex) {
               JOptionPane.showMessageDialog(desk, " Error Loading File" ,"Error",
                         JOptionPane.WARNING_MESSAGE);
               ais.close();
               throw ex;}
          } catch (Exception ex) {
               JOptionPane.showMessageDialog(desk, " Error Loading File" ,"Error",
                         JOptionPane.WARNING_MESSAGE);
          return result;
     }

You can find oracle documentation for sqlldr (11g) here:
http://docs.oracle.com/cd/B28359_01/server.111/b28319/ldr_concepts.htm#SUTIL003
If you have questions on details you should post your question in the special sqlldr forum:
Export/Import/SQL Loader & External Tables
hm

Pitch lowers on WAV files imported into Logic.

I'm working on a remix project for a friend, and every time I import a WAV file into Logic, the pitch lowers by about 10 percent. I checked the actual WAV itself by playing it through quicktime and itunes, and it's fine. It's only once I import it into Logic that I have issues. An ideas? ThankS!

Hi,
guess the imported Files are sampled with 48 kHz, while you are trying to import into a 44.1 kHz Project. Set the project sample rate to 48 kHz or convert the files down to 44.1 before importing
Fox

Masterd files back into wave burner

Hi everyone.
Im currently getting my projects mastered abroad; usually when I master locally the mastering engineer can embed the my ISRC codes and information on a CD ready for duplication. However I dont think I can get a CD sent to me, it would just be the mastered WAV file.
Can I take the mastered WAV file back into wave burner to insert the relevant information and codes and export it back out without effecting the file?

That doesn't sound like a problem to me as long as you don't add any plug-ins, normalize, or make any volume level changes. If the mastered files come back as individual tracks, you will have to deal with timing gaps between tracks. If they come back as a single file, you will need to insert track markers.
Out of curiosity, where do you have them mastered? Are you getting good results?

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.

How to overwrite the header part of a wave file?????

I need to overwirte the header part of a wave file. Any one has ideas or similar experiences?
basically, I want to continousely save wave data from a sound card. At the end of my program, I need to write the data size of the final wave file to the header part of the wave file.
Is there a function to replace the content of a file? Thanks

I assume that what you want to do is add documentation into the header of the wav file such as the media type, location, equipment used .....
Some one may have written some routines to do this in Labview, unfortunately all the ones I have are written in Visual Basic, quite a long time ago.
Here are some pointers in the event that no one has any Labview stuff, we could even knock a few togther, if its not been done.
The WAV file falls into a cateory of Windows files known as RIFF format files, in fact you will see this written into the file header almost at the very front.
I quote here from the Microsoft Technet Article:-
See here for a nice picture and here for full details, although for the header structure I prefer here.
"The basic building block of a RIFF file is a chunk. A chunk is
a logical unit of multimedia data, such as a single frame in a video
clip. Each chunk contains the following fields:
A four-character code specifying the chunk identifier
A doubleword value specifying the size of the data member in the chunk
A data field
Don't be put off by the fact that it often says AVI rather than WAV in the examples, the RIFF format is an extensible file format supporting lots of various media types even custom types. It's just that at the moment people are interested in Video.
There are standard fourcc codes that exist to hold various specific types of information and if one of these is not suitable you can create your own, there is (well was) even a way to regiser your own special type with Microsoft!!
There are a couple of ways of doing the job (assuming some kind sole does not donate a VI library).
1) Handle all the file i/o yourself by reading in the file and then insert the appropriate headers and rewrite the file out again.
Involved, but cross platform.
2) Muck about with Windows 'mmio' librarys (mmioCreateChunk).
Platform specific but possibly more reliable???
If you select option 1, then what you have to do is read in the appropriate chunk sizes(lengths) contained in the header, insert your appropriate FOURCC code, size(length) and data then adjust the affected 'chunk' size indicators to put all the various size indicators in the correct place, simple ehhh
If you select option 2, this requires linking to the appropriate DLL, it always my last resort to avoid DLL hell and portability issues.
As an example I wrote my own Turbo Pascal and later Visual Basic handlers which allowed attachment of lots of information from a form, and then created a special media format for the data to allow storing native measured data. These files are still recognised by WAV file readers today such as Media Player 10. Of course the data format is not understood because the encoding type was special, but that's the point.
One final point, there was a tool which read in a WAV file displaying all available headers, with C source (Windows 3.X) supplied in the MSDN toolkit. There are probalby other tools available by now, as a last resort If you decide to have a go I can send you my VB alpha version which reads all standard headers, it might write out a few as well (it was a very long time ago).
Good luck.
Message Edité par Conseils le 12-10-2005 04:09 PM

Mixing external wave file into video

I've recorded a video including sound of a mandolin being played. At the same time I recorded the sound with my external recorder which resulted in a wave file of better quality that the sound from the camera. I would like to import the video and the externally recorded wave file into a project, line up the external wave file with the sound recorded by the video camera and then delete the sound recorded with the video camera so I end up with a video that has the sound from the external recording system. Any recommendations on a good approach to accomplishing this task?

There's information here about how to use the new merged clips features in Premiere Pro CS5.5 to merge the audio from an external source with the video from the camera.

How do I convert an audio link into a wav file?

how do I convert an audio link into a wav file with my Imac?
A PC user showed me to right click an audio link where it asked if it wanted to save the link as a Wav file?
I saw this done in front of me. How do I do it with my Mac ?

This is how to do it in itunes http://support.apple.com/kb/ht1550
If I view the song in finder, all that i have to do is press command-i and it opens a window. In the section where you view the name and extension I took off the extension (.mp3 in my case) and replaced it with .wav

How can I convert an array into a .wav file?

I have an array of points that I need converted and written into a .wav file. Any help?

Indeed there is!
Look at the attached vi....that's about all there is to it!
Good Luck..
Eric
Eric P. Nichols
P.O. Box 56235
North Pole, AK 99705
Attachments:
ARRAY_TO_WAVE.vi ‏17 KB

How to Load a wav file into a JFrame

Ok, I have a JFrame with a JPanel and JButtons, play, pause, stop....etc...I have a menu with items load song and exit. Now when you click file load song, it pops up a JFileChooser.....that all works fine. Now I want to be able to select a wav file from that file chooser and have it load into the JFrame, displaying the name of the song in the JPanel...as a JLabel or whatever. Also how can I make the buttons, play pause stop, etc. work with the song. I.E. when i press play, the song plays, stop the song stops...any ideas?

Create a JLabel with the filename as its text. Add the JLabel onto your JPanel. Add the JPanel
onto the JFrame. Call setVisible(true) on the JFrame. For Java GUI basics, study:
http://java.sun.com/docs/books/tutorial/uiswing/index.html
For wav file playing:
String filename = "foo.wav";
java.applet.AudioClip clip
= java.applet.Applet.newAudioClip(new File(filename).toURI().toURL());
clip.play();

How do I import wav files from an external hard drive into itunes converting to aac?

I have 270 gig of wav files on an external hard drive. I am having trouble finding a way to import these into itunes while converting to aac

Note that .wav files do not carry a tag. If you import files of the form ..\<Artist>\<Album>\## <Name> into iTunes it will fill in the name property as the filename ## <Name> and all other fields will be blank. Assuming there is useful data in the path to capture, use Edit > Preferences > Avanced and disable the Keep... and Copy... options before adding the tracks to iTunes. You can then use the script TagFromFilename to fill in the blanks. Once this information is there you can convert the files to their new format and the converted files will inherit the track details.
You might want to consider initially converting to Apple Lossless which could halve the physical size of the collection with no loss of quality, and permanently embed the tag details. You can then choose to make a downsampled collection while retaining the full quality lossless copies and completely discarding the .wav originals.
tt2

"Input contract violation" error message when trying to import a .wav file into AE?

I am trying to import a .wav file into AE and keep getting an "input contract violation" message from the program followed by it shutting down. If it means anything, I bought the song online, but it has been paid for and I downloaded the .wav file after my purchase. Any idea of why this may be happening? Are there any particular settings which may be mandatory on the project timeline to import a 3min song? Thank you!

YOu have not provided any system info, not told us what version of AE you use, your audio setting, what specifics in the audio file and a million other things. See how it may not be possible for anyone to tell you anything beyond "Your computter sucks!" ???
Mylenium

Logic Won't Let Me 'Drop' .Wav Files Into A Session??

Hi all,
Trying to simply drag and drop .wav files from a folder on an external hard drive into an empty Logic Pro 9 session. The audio tracks are stereo (have also tried mono) and the input/outputs seem to be fine. The session is also saved to the external. I am not using an external audio interface.
I have also tried importing using the 'File-->import' commands and this allows me to import one file at a time, but not in bulk. This seems odd as I could've sworn I didn't have this problem in past attempts.
Let me know your thoughts please!
Ps. Logic rules, but I'm very tempted to get acquainted with Pro Tools!

GTBannah wrote:
Are you sure they are audio tracks, and not MIDI or Instrument tracks?
Ha ha... Yes I am sure. A good thought, though I did double checked this before posting the problem.
Could it be something to do with the sample rate? Ie. the audio files were recorded at a certain sample rate and now I need to make it match with the sample rate in Logic?

Why is it that when I saved the waveform of my acquired signal into a wav file, the output waveform of my saved file turn out to be different from my input signal?

Hi all,
I am currently doing a heart sound analyzer project. I acquired signal using NI USB 6008 continuosly. My program is able to saved the acquired signal into a wave file. But when I browse open the saved file, the output waveform did not look the same as the acquired input signal and also the sound of the heartbeat captured cannot be heard from the loud speaker though I have saved the file as a wav file. I am not sure if the method i use to save the acquired data as a wav file is correct. I have attach the print screen of the differences of the output I have notice and the program I have created. I would be glad if anyone here is able to troubleshoot my program.
Thanks.
Attachments:
heart sound analyzer.zip ‏257 KB

Hi Brett Burger,
Thanks for your reply. For your information, I have set the sampling rate as 10000 as for the sound format, I have set the bits per sample as 16 bit, the rate as 11025 and the sound quality as mono. I tried using your method by changing the sampling rate as 8K but still my program encounter the same problem.
I wish to also create a button that is able to generate a preformatted report that contains VI documentation, data the VI returns, and report properties, such as the author, company, and number of pages only when I click on the button. I have created this in my program, but I am not sure why is it not working. Can you help troubleshoot my program. Or do you have any samples to provide me. Hope to hear from you soon.
Many thanks.
Regards,
min
Attachments:
Heart Sounds1.vi ‏971 KB

Bytes array into a wav file

hi every1, any1 can help plz, how to write bytes from an audio file (which is stored in an array) back into a wave file thanx

Hi,
it is something like this:
byte[] myByteArray;          //your bytearray
AudioFormat myAudioFormat;     //neccessary to interpret the bytes in the right manner
long mySampleFrameLength;     //how long your audiodata is
AudioInputStream ais = new AudioInputStream(new ByteArrayInputStream(myByteArray), myAudioFormat, mySampleFrameLength);
AudioSystem.write(ais, AudioFileFormat.Type.WAVE, new File("myWaveFile.wav"));

WAV File Soundkits into GB.,...

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