How to play other audio files in Podcasts app?

Hi all. Whenever I add audiobooks or recordings of radio shows to my iPhone, they go into the Music app. However, this doesn't have a placeholder function (that is, it doesn't remember at which you point you paused the file) or a sleep timer like the Podcasts app does. I find this really annoying, especially the lack of placeholder. Is there any way to get these things to play from Podcasts instead? I've searched on this but can't find any other questions about it. Surely other people find this irritating as well? Hoping there is some solution, maybe using a third party app?

Close Windows Media Player. Make sure you know where the song files are on your hard drive. Use the iTunes command File > Add File (or Folder) to Library.
If they are MP3s they will add directly to your iTunes library. If they are non-protected WMA files, they will be automatically converted and added. If they are something else, "measures" must be taken...
Let is know how this works!

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    import sun.audio.*; //import the sun.audio package
    import java.io.*;
    //** add this into your application code as appropriate
    // Open an input stream to the audio file.
    InputStream in = new FileInputStream(Filename);
    // Create an AudioStream object from the input stream.
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    Hello to everybody,
    I am working on a application that is supposed to manage audio file. At the moment I 'm using the SourceDataLine instead of the Clip. But I've got a problem. When the user pushes the Stop button I want that if he pushes Play again the track starts from the begin. I've used the drain() and flush() method but it doesn't work because the track resumes from the point when has been interrupted. Here part of the code:
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    mauriziodvt wrote:
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    import sun.audio.*; //import the sun.audio package
    import java.io.*;
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    Hi Ibrahim,
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    http://www.deansdirectortutorials.com/
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    package com.d2e.MyPlugin;
    import com.servoy.j2db.scripting.IScriptObject;
    import javax.swing.*;
    import java.awt.*;
    import java.awt.event.*;
    import java.io.*;
    import java.awt.Button;
    import javax.sound.sampled.*;
    import org.xiph.speex.*;
    import org.xiph.speex.spi.*;
    import  sun.audio.*;    //import the sun.audio package
    public class MyPluginProvider implements IScriptObject  {
          public void JSencode()
             throws IOException{
               /** Version of the Speex Encoder */
                final String VERSION = "Java Speex Command Line Encoder v0.9.7 ($Revision: 1.5 $)";
                /** Copyright display String */
                final String COPYRIGHT = "Copyright (C) 2002-2004 Wimba S.A.";
                /** Print level for messages : Print debug information */
                final int DEBUG = 0;
                /** Print level for messages : Print basic information */
                final int INFO  = 1;
                /** Print level for messages : Print only warnings and errors */
                final int WARN  = 2;
                /** Print level for messages : Print only errors */
                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
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                 //  fileType = AudioFileFormat.Type.AIFF;
                 //  audioFile = new File("junk.aif");
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                 //  fileType = AudioFileFormat.Type.AU;
                 //  audioFile = new File("junk.au");
                // }else if(sndBtn.isSelected()){
                //   fileType = AudioFileFormat.Type.SND;
                 //  audioFile = new File("junk.snd");
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                   targetDataLine.open(audioFormat);
                   targetDataLine.start();
                   AudioSystem.write(
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                         fileType,
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    Ah, those scroll bars. They don't stand out, as much, as I would like. I always overlook the ones in the Share Panel, though I have used them many times.
    Unfortunately, PrE has chosen an arrangement, where Audio & Video Tracks are grouped. I like the PrPro layout, where all Video Tracks are above a "centerline" and all Audio Tracks are below that.
    About the closest that one can come in PrE is to use Delete Unused Tracks, to remove empty Video Tracks, or to View Video Tracks (alone), or View Audio Tracks (alone). With a combo of those, one can customize the Timeline a bit.
    When cleaned up, one can then Add Tracks, for any additional Audio Tracks necessary.
    If one did ONLY the View Audio, they loose the ability to refer to the Video Track, and with SFX, that is necessary, at least for me.
    I like to keep each type of Audio on its own Track. That makes actions, like applying Track Keyframes, or adjustments to JUST that Clip, or Clip type. If I have SFX, music, camera Audio, and other Audio sources, I will put many SFX on their own Audio Track, the camera Audio on another, and then the music one one, or more additional Audio Tracks. Not too long ago, I had one Project with about 10 separate SFX Audio Tracks, one for a "loon," one for "crickets," three for "frogs," one for "pond ambiance," and then two for music (as I was doing faux DD 5.1 SS in PrPro), with one from Music F (front) and Music R (rear).
    Good luck, and happy editing,
    Hunt

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