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AudioInputStream from file conversion gives "frameLength=-1"

I am trying to obtain an AudioInputStream from a file in a specific format. When I open the file in fileStream I get a positive frameLength meaning the file can be read and the AudioInputStream effectively has data. Now I want to convert it with AudioSystem with AudioSystem.getAudioInputStream(format, fileStream); and the obtained stream audioStream has a frameLength of -1. I even made sure the conversion is supported and it sure is.
public void openFile() throws IOException, UnsupportedAudioFileException, LineUnavailableException {
AudioInputStream fileStream = AudioSystem.getAudioInputStream(audioFile);
AudioFormat format = new AudioFormat(sampleRate, 8, 1, true, true);
boolean supported = AudioSystem.isConversionSupported(format, fileStream.getFormat());
if(supported) {
audioStream = AudioSystem.getAudioInputStream(format, fileStream);
System.out.println("Opened file: " + audioFile.getName());
}else{
System.out.println("Couldn't open file: " + audioFile.getName());
throw new IOException();
}
}
I am completely stuck and I haven't found anyone with the same exact problem. I welcome any suggestions of different libraries but I would prefer to keep using this one as I'm used to it.

Okay figured it out, I'm dumb.
A frameLength of -1 is, according to the AudioSystem constants, NOT_SPECIFIED which means that, when converting audio, you lose that information.
I was thinking that was the problem because of another unrelated problem.

How to get panning control forma a .mp3 file?

I´m learning how to play sound in java but with advanced controls.
I´ve found one problem: The javax.sound.sampled.AudioInputStream doesn´t support Mp3 files, and i´m running out of ideas to find how to get control of panning.
I managed to play an Mp3 file using javazoom.jl.player.advanced.AdvancedPlayer, but it doesn´t have a panning control, or i haven´t founded it.
My actual code opens a file, if the format is compatible with AudioInputStream, it plays only the right channel. If the format doesn´t, it plays using AdvancedPlayer.
Do you know a way to get panning control of mp3 files?
My code here:
import javazoom.jl.decoder.JavaLayerException;
import javazoom.jl.player.advanced.AdvancedPlayer;
import javax.sound.sampled.*;
import javax.swing.*;
import java.io.File;
import java.io.FileInputStream;
import java.io.IOException;
public class AudioPlayerExample2 {
private static final int BUFFER_SIZE = 4096;
public static void main(String[] args) throws IOException, LineUnavailableException, JavaLayerException {
JFileChooser fileChooser = new JFileChooser();
fileChooser.showOpenDialog(null);
new AudioPlayerExample2().play(fileChooser.getSelectedFile());
}
void play(File file) throws IOException, LineUnavailableException, JavaLayerException {
AudioInputStream audioStream;
try {
audioStream = AudioSystem.getAudioInputStream(file);
AudioFormat format = audioStream.getFormat();
System.err.println(format.toString());
DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
SourceDataLine audioLine = (SourceDataLine) AudioSystem.getLine(info);
audioLine.open(format);
audioLine.start();
FloatControl pan = (FloatControl) audioLine.getControl(FloatControl.Type.PAN);
byte[] bytesBuffer = new byte[BUFFER_SIZE];
int bytesRead = -1;
while ((bytesRead = audioStream.read(bytesBuffer)) != -1) {
pan.setValue((float) (1));
audioLine.write(bytesBuffer, 0, bytesRead);
}
audioLine.drain();
audioLine.close();
audioStream.close();
} catch (UnsupportedAudioFileException e) {
FileInputStream fis = new FileInputStream(file);
AdvancedPlayer player = new AdvancedPlayer(fis);
player.play();
}
}
}

Panning and volume controls are system dependent and can sometimes be a bit flaky even if they are in place. For example, if you change the volume or pan setting too much at once, the discontinuity causes a click.
One solution is to go in there on a per-frame basis and make the changes yourself. For example, see "Manipulating the Audio Data Directly" at the end of the tutorial Processing Audio with Controls.
For an example, check out the code from the next tutorial on the trail: Using Files and Format Converters. Look under the heading "Reading Sound Files" and look for the comment in the code "\ Here, do something useful..."
I invite you to also take a look at the code I wrote and have made available, a class called AudioCue that has real time panning as well as real time volume and pitch playback controls. I've added smoothing (1024 steps for panning changes) to help mitigate the possibility of discontinuities.
It will be up to you to take the mp3 file and decode it into an array of audio data. I think that the javazoom libraries made available on github should give you enough code access to figure out how to do this (I did it for ogg/vorbis decoding). Once you have a float array of audio data (stereo, signed floats ranging from -1 to 1), this can be directly loaded into AudioCue.

First of all, thanks to Andrew Thompson and Phil Freihofner, I feel very good about being part of this community and having someone to trust. You really make feel happy :)
I leave here the full code that does exactly what I wanted.
As the JavaZoom MP3 SPI Documentation says: Make sure that JLayer, Tritonus and MP3SPI librairies are available in your CLASSPATH.
import javax.sound.sampled.*;
import javax.swing.*;
import java.io.File;
import java.io.IOException;
public class Test {
public static void main(String[] args) throws IOException,
UnsupportedAudioFileException, LineUnavailableException {
JFileChooser chooser = new JFileChooser();
chooser.showOpenDialog(null);
String path = chooser.getSelectedFile().getAbsolutePath();
System.err.println(path);
File file = new File(path);
AudioInputStream baseStream = AudioSystem.getAudioInputStream(file);
AudioFormat baseFormat = baseStream.getFormat();
System.err.println(baseFormat.toString());
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
baseFormat.getSampleRate(),
16, baseFormat.getChannels(), baseFormat.getChannels() * 2,
baseFormat.getSampleRate(), true);
DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
AudioInputStream stream = AudioSystem.getAudioInputStream(format, baseStream);
SourceDataLine audioLine = (SourceDataLine) AudioSystem.getLine(info);
audioLine.open(format);
audioLine.start();
FloatControl pan = (FloatControl) audioLine.getControl(FloatControl.Type.PAN);
pan.setValue(1);
int BUFFER_SIZE = 4096;
byte[] buffer = new byte[BUFFER_SIZE];
int read = -1;
while((read = stream.read(buffer)) != -1){
audioLine.write(buffer, 0, read);
}
audioLine.drain();
audioLine.close();
}
}

PDFBox creating Sound object with link/reference to external mp3 or wav file

I am writing a utility application using open source java based PDFBox to convert PDF file containing 'Hyperlink to open an mp3 file' to replace it with sound object.
I used PDFBox API since it appears to be mature enough to work with Sound object. I could read the PDF file and find the hyperlink with reference to mp3. But I am not able to replace it with sound object. I created the Sound Object and associate with action but it does not work. I think I am missing some important part how to create Sound object using PDActionSound object. Is it possible to refer to external wav file using PDFBox API?
for (PDPage pdPage : pages) {
List<PDAnnotation> annotations = pdPage.getAnnotations();
for (PDAnnotation pdAnnotation : annotations) {
if (pdAnnotation instanceof PDAnnotationLink) {
PDAnnotationLink link = ((PDAnnotationLink) pdAnnotation);
PDAction action = link.getAction();
if (action instanceof PDActionLaunch) {
PDActionLaunch launch = ((PDActionLaunch) action);
String fileInfo = launch.getFile().getFile();
if (fileInfo.contains(".mp3")) {
/* create Sound object referring to external mp3*/
//something like
PDActionSound actionSound = new PDActionSound(
soundStream);
//set the ActionSound to the link.
link.setAction(actionSound);
}
}
}
}
}
How to create sound object (PDActionSound) and add to link successfully?

Speaking of mature, that part has never been used, and now that I had a closer look at the code, I think some work remains to be done... Please try this, I created this with PDFBox 2.0 after reading the PDF specification:
PDSimpleFileSpecification fileSpec = new PDSimpleFileSpecification(new COSString("/C/dir1/dir2/blah.mp3")); // see "File Specification Strings" in PDF spec
COSStream soundStream = new COSStream();
soundStream.createOutputStream().close();
soundStream.setItem(COSName.F, fileSpec);
soundStream.setInt(COSName.R, 44100); // put actual sample rate here
PDActionSound actionSound = new PDActionSound();
actionSound.getCOSObject().setItem(COSName.getPDFName("Sound"), soundStream));
link.setAction(actionSound); // reassign the new action to the link annotation
edit: as the above didn't work, here's an alternative solution as requested in the comments. The file is embedded. It works only with .WAV files, and you have to know details of them. About 1/2 seconds are lost at the beginning. The sound you should hear is "I am Al Bundy". I tried with MP3 and didn't succeed. While googling, I found some texts saying that only "old" formats (wav, aif etc) are supported. I did find another way to play sounds ("Renditions") that even worked with embedded mp3 in another product, but the generated structure in the PDF is even more complex.
COSStream soundStream = new COSStream();
OutputStream os = soundStream.createOutputStream(COSName.FLATE_DECODE);
URL url = new URL("http://cd.textfiles.com/hackchronii/WAV/ALBUNDY1.WAV");
InputStream is = url.openStream();
// FileInputStream is = new FileInputStream(".....WAV");
IOUtils.copy(is, os);
is.close();
os.close();
// See p. 506 in PDF spec, Table 294
soundStream.setInt(COSName.C, 1); // channels
soundStream.setInt(COSName.R, 22050); // sampling rate
//soundStream.setString(COSName.E, "Signed"); // The encoding format for the sample data
soundStream.setInt(COSName.B, 8); // The number of bits per sample value per channel. Default value: 8
// soundStream.setName(COSName.CO, "MP3"); // doesn't work
PDActionSound actionSound = new PDActionSound();
actionSound.getCOSObject().setItem(COSName.getPDFName("Sound"), soundStream);
link.setAction(actionSound);
Update 9.7.2016:
We discussed this on the PDFBox mailing list, and thanks to Gilad Denneboom we know two more things:
1) in Adobe Acrobat it only lets you select either WAV or AIF files
2) code by Gilad Denneboom with MP3SPI to convert MP3 to raw:
private static InputStream getAudioStream(String filename) throws Exception {
File file = new File(filename);
AudioInputStream in = AudioSystem.getAudioInputStream(file);
AudioFormat baseFormat = in.getFormat();
AudioFormat decodedFormat = new AudioFormat(
AudioFormat.Encoding.PCM_UNSIGNED,
baseFormat.getSampleRate(),
baseFormat.getSampleSizeInBits(),
baseFormat.getChannels(),
baseFormat.getChannels(),
baseFormat.getSampleRate(),
false);
return AudioSystem.getAudioInputStream(decodedFormat, in);
}

Concatenating WAV files in Java

Here is my code that concatenates four wav files and produces wavAppended.wav. This concatenated file nicely plays in Windows Media Player.
But through the PlaySound class, only the one.wav can be heard.
Can anyone help?
class PlaySound extends Object implements LineListener
{
File soundFile;
JDialog playingDialog;
Clip clip;
public void PlaySnd(String s) throws Exception
{
JFileChooser chooser = new JFileChooser();
soundFile = new File(s);
Line.Info linfo = new Line.Info(Clip.class);
Line line = AudioSystem.getLine(linfo);
clip = (Clip) line;
clip.addLineListener(this);
AudioInputStream ais = AudioSystem.getAudioInputStream(soundFile);
clip.open(ais);
clip.start();
}
public void update(LineEvent le)
{
LineEvent.Type type = le.getType();
playingDialog.setVisible(false);
clip.stop();
clip.close();
}
}
public class Main
{
public static void main(String[] args)
{
int i;
String wavFile[] = new String[4];
wavFile[0] = "D://one.wav";
wavFile[1] = "D://two.wav";
wavFile[2] = "D://three.wav";
wavFile[3] = "D://space.au";
AudioInputStream appendedFiles;
try
{
AudioInputStream clip0=AudioSystem.getAudioInputStream(new File(wavFile[0]));
AudioInputStream clip1=AudioSystem.getAudioInputStream(new File(wavFile[1]));
AudioInputStream clip3;
for (i=0;i<4;i++)
{
appendedFiles = new AudioInputStream(
new SequenceInputStream(clip0, clip1),
clip0.getFormat(),
clip0.getFrameLength() + clip1.getFrameLength());
AudioSystem.write(appendedFiles, AudioFileFormat.Type.WAVE, new File("D:\\wavAppended.wav"));
clip3 = AudioSystem.getAudioInputStream(new File("D:\\wavAppended.wav"));
clip0=clip3;
clip1 = AudioSystem.getAudioInputStream(new File(wavFile[i+2]));
}
PlaySound p = new PlaySound();
p.PlaySnd("D://wavAppended.wav");
}
catch (Exception e)
{
e.printStackTrace();
}
}
}

WAV files don't work that way -- you can't just throw multiple files together (same as you can't concatenate JPEG images, for instance), as there's a header on the data, and there are multiple different formats the data may be in. I'm surprised that the file loads at all.

To get you started with the WAV processing you may have a look at my small project. It can copy and paste WAV files together based on an time index file. The project should contain all the Java WAV processing you need (using javax.sound.sampled). The Butcher implementation and Composer contain the actual processing.
The idea is simple: take input audio files and create a index of words
contained in these files. The index entry is the word, start time and
end time. When a new sentence is created it will be stitched together
with single words taken from the index.
The AudioInputStream is the main class to interact with the Java Sound
API. You read audio data from it. If you create audio data you do this
by creating a AudioInputStream the AudioSystem can read from. The
actual encoding is done by the AudioSystem implementation depending on
the output audio format.
The Butcher class is the one concerned with audio files. It can read
and write audio files and create AudioInputStreams from an input byte
array. The other interesting think the Butcher can is cutting samples
from a AudioInputStream. The AudioInputStream consists of frames that
represent the samples of the PCM signal. Frames have a length of
multiple bytes. To cut a valid range of frames from the
AudioInputStream one has to take the frame size into account. The
start and end time in milliseconds have to be translated to start byte
and end bytes of the start frame and end frame. (The start and end
data is stored as timestamps to keep them independent from the
underlying encoding of the file used.)
The Composer creates the output file. For a given sentence it takes
the audio data for each word from the input files, concatenates the
audio data and writes the result to disk.
In the end you'll need some understanding of the PCM and the WAV format. The Java sound API does not abstract that away.

In above given example you need to use the SequenceInputStream then it will work fine. please find my code below to join two files.
import java.io.File;
import java.io.IOException;
import java.io.SequenceInputStream;
import javax.sound.sampled.AudioFileFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
public class JoinWav{
public static void main(String... args) throws Exception{
String wav_1 = "1497434542598100215.wav";
String wav_2 = "104860397153760.wav";
AudioInputStream stream_1 = AudioSystem.getAudioInputStream(new File(wav_1));
AudioInputStream stream_2 = AudioSystem.getAudioInputStream(new File(wav_2));
System.out.println("Info : Format ["+stream_1.getFormat()+"] Frame Length ["+stream_1.getFrameLength()+"]");
AudioInputStream stream_join = new AudioInputStream(new SequenceInputStream(stream_1,stream_2),stream_1.getFormat(),stream_1.getFrameLength()+stream_2.getFrameLength());
AudioSystem.write(stream_join,AudioFileFormat.Type.WAVE,new File("join.wav"));
}
}

java.io.IOException: mark/reset not supported

try {
//String location = dir1.getCanonicalPath()+"\\app_yamb_test1\\mySound.au";
//displayMessage(location);
AudioInputStream audio2 = AudioSystem.getAudioInputStream(getClass().getResourceAsStream("mySound.au"));
Clip clip2 = AudioSystem.getClip();
clip2.open(audio2);
clip2.start();
} catch (UnsupportedAudioFileException uae) {
System.out.println(uae);
JOptionPane.showMessageDialog(null, uae.toString());
} catch (IOException ioe) {
System.out.println("Couldn't find it");
JOptionPane.showMessageDialog(null, ioe.toString());
} catch (LineUnavailableException lua) {
System.out.println(lua);
JOptionPane.showMessageDialog(null, lua.toString());
}
This code works fine when I run the application from netbeans. The sound plays and there are no exceptions. However, when I run it from the dist folder, the sound does not play and I get the java.io.IOException: mark/reset not supported in my message dialog.
How can I fix this?

The documentation for AudioSystem.getAudioInputStream(InputStream) says:
The implementation of this method may
require multiple parsers to examine
the stream to determine whether they
support it. These parsers must be able
to mark the stream, read enough data
to determine whether they support the
stream, and, if not, reset the
stream's read pointer to its original
position. If the input stream does not
support these operation, this method
may fail with an IOException.
Therefore, the stream you provide to this method must support the optional mark/reset functionality. Decorate your resource stream with a BufferedInputStream.
//read audio data from whatever source (file/classloader/etc.)
InputStream audioSrc = getClass().getResourceAsStream("mySound.au");
//add buffer for mark/reset support
InputStream bufferedIn = new BufferedInputStream(audioSrc);
AudioInputStream audioStream = AudioSystem.getAudioInputStream(bufferedIn);

After floundering about for a while and referencing this page many times, I stumbled across this which helped me with my problem. I was initially able to load a wav file, but subsequently only could play it once, because it could not rewind it due to the "mark/reset not supported" error. It was maddening.
The linked code reads an AudioInputStream from a file, then puts the AudioInputStream into a BufferedInputStream, then puts that back into the AudioInputStream like so:
audioInputStream = AudioSystem.getAudioInputStream(new File(filename));
BufferedInputStream bufferedInputStream = new BufferedInputStream(audioInputStream);
audioInputStream = new AudioInputStream(bufferedInputStream, audioInputStream.getFormat(), audioInputStream.getFrameLength());
And then finally it converts the read data to a PCM encoding:
audioInputStream = convertToPCM(audioInputStream);
With convertToPCM defined as:
private static AudioInputStream convertToPCM(AudioInputStream audioInputStream)
{
AudioFormat m_format = audioInputStream.getFormat();
if ((m_format.getEncoding() != AudioFormat.Encoding.PCM_SIGNED) &&
(m_format.getEncoding() != AudioFormat.Encoding.PCM_UNSIGNED))
{
AudioFormat targetFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
m_format.getSampleRate(), 16,
m_format.getChannels(), m_format.getChannels() * 2,
m_format.getSampleRate(), m_format.isBigEndian());
audioInputStream = AudioSystem.getAudioInputStream(targetFormat, audioInputStream);
}
return audioInputStream;
}
I believe they do this because BufferedInputStream handles mark/reset better than audioInputStream. Hope this helps somebody out there.

Just came across this question from someone else with the same problem who referenced it. Looks like this issue arose with Java 7.
Oracle Bug database, #7095006
A test, executed when InputStream is the argument to the getAudioInputStream() method, is triggering the error. The existence of Mark/Reset capabilities in the audio resource file have no bearing on whether the Clip will load and play. Given that, there is no reason to prefer an InputStream as the argument, when a URL or File suffice.
If we substitute a URL as the argument, this needless test is not executed. Revising the OP code:
AudioInputStream ais = AudioSystem.getAudioInputStream(getClass().getResource(fileName));
Details can be seen in the API, in the description text for the two forms.
AudioSystem.getAudioInputStream(InputStream)
AudioSystem.getAudioInputStream(URL)

The problem is that you're input stream has to support the methods mark and reset. At least if mark is supported you can test with: AudioInputStream#markSupported.
So you should maybe use a different InputStream.