Audio files from analog 4 track out of sync

After many years I've finally decided to transfer my old 4 track demos from cassette to my Mac.
I rented a 4 track recorder to play the tapes. I can only transfer 2 tracks at a time, so I have 2 sets 2 tracks each, for a total of 4 tracks. (still with me?)
For some reason, the different sets play at different speeds. So if everything is at sync at the beginning of the song, they're well out of beat 90 seconds later.
How can I speed up or slow down a single track? All of the data is in GB, so it's hard to get out.
Can I do this in Logic Express? (a program which I own but have no idea how to use, since life and work and motorcycles and kids get in the way of playing too much on the Mac...
20" iMac G5 2.0/MacBook 1.83/eMac 1.25   Mac OS X (10.4.8)  

Export each track as a separate AIFF file.
Open both tracks in AmadeusPro. Copy and paste into a new file with two tracks.
Select one track and use Effects>Change Pitch and Speed. This will provide incremental adjustments till both waveforms match up. But remember, you need to adjust both pitch AND speed together to reproduce a mechanical speed difference.
Once you have both tracks running in sync, use this opportunity to separate your two tracks into four by selecting Sound>Split Track. Now you have 4 separate tracks that are now in sync.
Export each track as AIFF files and reimport back to GarageBand by dragging them onto the GarageBand timeline.

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    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
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          public void JSencode()
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               /** 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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                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
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               dis.readFully(temp, 0, HEADERSIZE+4);
               // Read other header chunks
               dis.readFully(temp, 0, HEADERSIZE);
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               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) {
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                   channels = readShort(temp, 2);
                   sampleRate = readInt(temp, 4);
                   if (readShort(temp, 14) != 16) {
                     System.err.println("Not a 16 bit file " + readShort(temp, 18));
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                   // 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).
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            * @param offset the offset from which to start reading.
            * @return the integer value of the reassembled bytes.
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         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)
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                  // 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;
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        private void captureAudio(){
          try{
            //Get things set up for capture
            audioFormat = getAudioFormat();
            DataLine.Info dataLineInfo =
                                new DataLine.Info(
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                                  audioFormat);
            targetDataLine = (TargetDataLine)
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            // 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
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        // well for you, try some of the other
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          float sampleRate = 44100.0F;
          //8000,11025,16000,22050,44100
          int sampleSizeInBits = 16;
          //8,16
          int channels = 1;
          //1,2
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          //true,false
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          //true,false
          return new AudioFormat(sampleRate,
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                                 signed,
                                 bigEndian);
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               public void run(){
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                 File audioFile = null;
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                 audioFile = new File("junk.wav");
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                   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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    Thank you for using the Apple Support Communities
    You can synchronize audio files from iTunes to your iPod.
    https://www.apple.com/support/ipodclassic/how-to/
    -Zeph

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    If the web page is designed to let you save files it will often have a download link, or if it is a link to the file directly you can try to right click on the file.  If it isn't set up that way then they probably don't intend for the file to be downloaded.

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    Hi,
    Question:
    1. You have a folder for each DVD whose video contents you edited.
    Answer: Yes
    Question:
    You talk about projects 34, 35, 36, 37. Are these projects associated with only one specific DVD?
    What is supposed to be in each project - just edited VTS_01_1.VOB or edits from the others in the series
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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
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    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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    Click on all of the albums, artists, playlists that you want to sync
    Click on the Sync Music Heading.
    Click on Apply in the lower right corner of iTunes
    Follow the same procedure for Movies.

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    iTunes on the PC will convert wma. iTunes on a Mac won't.
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