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Can't hear sound from bluetooth speaker

I'm using Raspberry Pi with Ubuntu 22.10 and Java 19.
I'm trying to play a sound with the following code:
private static void testMixers() throws IOException, UnsupportedAudioFileException
{
try (AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(Objects.requireNonNull(Listener.class.getResourceAsStream("/alarmSound.wav")))))
{
final Scanner scanner = new Scanner(System.in);
for (Mixer.Info info : AudioSystem.getMixerInfo())
{
System.out.println("Testing sound for mixer \"" + info + "\"...");
try (Clip clip = AudioSystem.getClip(info))
{
clip.open(audioInputStream);
clip.setFramePosition(0);
clip.start();
System.out.println("Did you hear any sound? (true|false)");
if (Boolean.parseBoolean(scanner.nextLine()))
return;
}
catch (Exception e)
{
System.out.println(e);
}
}
}
System.out.println("Mixer not found!");
}
alarmSound.wav is under src/main/resources (this is a Maven project).
AudioSystem.getMixerInfo() gives me the following Mixer.Info[]:
Port Headphones [hw:0], version 5.19.0-1006-raspi
Port vc4hdmi0 [hw:1], version 5.19.0-1006-raspi
Port vc4hdmi1 [hw:2], version 5.19.0-1006-raspi
Headphones [default], version 5.19.0-1006-raspi
Headphones [plughw:0,0], version 5.19.0-1006-raspi
vc4hdmi0 [plughw:1,0], version 5.19.0-1006-raspi
vc4hdmi1 [plughw:2,0], version 5.19.0-1006-raspi
Only the Headphones Mixers don't give the error:
java.lang.IllegalArgumentException: Line unsupported: interface Clip supporting format PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian
I didn't tested the headphone jack.
This is completely different from Ubuntu 22.04 which gave more Mixer.Infos.
Is there any workaround? it seems that every Ubuntu version requires different workaround for being able to play a sound that can be heard not through the headphones (but from bluetooth speakers or via HDMI for example).

Unwanted downsampling : Java Sound

I have been trying to manually read a wav file in Java and read an array of bytes then write to an audio buffer for playback. I am receiving playback but it is heavily distorted. Java sound supports 16 bit sample rates but not 24-bit.
I went in to Logic 9 and exported a 24-bit audio file in to 16-bit and then used with my program. Originally, the 24-bit samples would produces white noise. Now I can hear my sample but very distorted and sounds like it has been bit crushed.
Can anyone help me to get a clean signal?
I am very new to audio programming but I am currently working on a basic Digital Audio Workstation.
import javax.sound.sampled.*;
import javax.sound.sampled.DataLine.Info;
import javax.swing.filechooser.FileNameExtensionFilter;
import java.io.*;
public class AudioData {
private String filepath;
private String filepath1;
private File file;
private byte [] fileContent;
private Mixer mixer;
private Mixer.Info[] mixInfos;
private AudioInputStream input;
private ByteArrayOutputStream byteoutput;
public static void main (String [] args) {
AudioData audiodata = new AudioData();
}
public AudioData () {
filepath = "/Users/ivaannagen/Documents/Samples/Engineering Samples - Obscure Techno Vol 3 (WAV)/ES_OT3_Kit03_Gmin_130bpm/ES_OT3_Kit03_FX_Fast_Snare_Riser_Gmin_130bpm.wav";
filepath1 = "/Users/ivaannagen/Documents/Samples/dawsampletest.wav";
file = new File (filepath1);
readAudio();
}
public void readAudio () {
mixInfos = AudioSystem.getMixerInfo();
mixer = AudioSystem.getMixer(mixInfos[0]);
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, false);
// set up an audio format.
try {
DataLine.Info info = new DataLine.Info(SourceDataLine.class, format); // creates data line with class type and audio format.
SourceDataLine source = (SourceDataLine) AudioSystem.getLine(info);
System.out.println("Size of data line buffer: " + source.getBufferSize());
fileContent = new byte [source.getBufferSize() / 50];
byteoutput = new ByteArrayOutputStream();
input = AudioSystem.getAudioInputStream(file);
int readBytes = 0;
while ((readBytes = input.read(fileContent, 0, fileContent.length)) != -1) {
byteoutput.write(fileContent, 0, readBytes);
}
System.out.println("Size of audio buffer: " + fileContent.length);
//byteoutput.write(0);
// byteoutput.write(0);
System.out.println("Size of audio buffer: " + byteoutput.size());
source.open(format, source.getBufferSize()); // line must be open to be recognised by the mixer.
Line[] lines = mixer.getSourceLines();
System.out.println("mixer lines: " + lines.length);
// for(byte bytes: fileContent) {
// System.out.println(bytes);
// }
Thread playback = new Thread () {
public void run () {
// System.out.println((byteoutput.size() +2) % 4);
source.start(); // play (buffer originally empty)
source.write(byteoutput.toByteArray(), 0, byteoutput.size()); // write input bytes to output buffer
} // end run (to do).
}; // end thread action
playback.start(); // start thread
}
catch (LineUnavailableException lue) {
System.out.println(lue.getMessage());
}
catch (FileNotFoundException fnfe) {
System.out.println(fnfe.getMessage());
}
catch(IOException ioe) {
System.out.println(ioe.getMessage());
}
catch(UnsupportedAudioFileException uafe) {
System.out.println(uafe.getMessage());
}
}
}

Whether or not you can load and play a 24-bit file is system dependent, afaik.
I use Audacity for conversions. You should be able import your file into Audacity and export it as 16-bit, stereo, little-endian, 44100 fps, and then load that export with Java's AudioInputStream.
What you hear when playing from Audacity or from Java should be pretty much identical (adjusting for volume). If not, the most likely reason probably pertains to a mistake or overlook in the code, which is very easy to do.
The use of a ByteOutputStream in your code is superfluous. Read from the AudioInputStream into a fixed-size byte array (size being the buffer length, I recommend trying 8 or 16 * 1024 bytes as a first try) and then use the SourceDataLine write method to ship that array.
Following is code that works on my system for loading a playing a "CD Quality" wav called "a3.wav" that I have that is in the same directory as the Java class. You should be able to swap in your own 44100, 16-bit, stereo, little-endian wav file.
I've commented out an attempt to load and play a 24-bit wav file called "spoken8000_24.wav". That attempt gave me an IllegalArgumentException: No line matching interface SourceDataLine supporting format PCM_SIGNED 8000.0 Hz, 24 bit, stereo, 6 bytes/frame, little-endian is supported.
I have to admit, I'm unclear if my system doesn't provide the needed line or if I might have coded the format incorrectly! My OS can certainly play the file. So I'm thinking there is a distinction between what an OS can do and what a "Mixer" on a given system provides to Java.
As a get-around, I just always convert everything to "CD Quality" format, as that seems to be the most widely supported.
public class TriggerSound_SDL extends JFrame
{
public TriggerSound_SDL()
{
JButton button = new JButton("Play Sound");
button.addActionListener(e -> new Thread(() -> playBuzzer()).start());
getContentPane().add(button);
}
private void playBuzzer()
{
try
{
URL url;
url = getClass().getResource("a3.wav");
// url = getClass().getResource("spoken8000_24.wav");
AudioInputStream ais = AudioSystem.getAudioInputStream(url);
System.out.println(ais.getFormat());
AudioFormat audioFmt;
// "CD Quality" 44100 fps, 16-bit, stereo, little endian
audioFmt = new AudioFormat(
AudioFormat.Encoding.PCM_SIGNED,
44100, 16, 2, 4, 44100, false);
// 8000 fps, 32-bit, stereo
// audioFmt = new AudioFormat(
// AudioFormat.Encoding.PCM_SIGNED,
// 8000, 24, 2, 6, 8000, false);
Info info = new DataLine.Info(SourceDataLine.class,
audioFmt);
SourceDataLine sdl = (SourceDataLine)AudioSystem.getLine(info);
int bufferSize = 16 * 1024;
byte[] buffer = new byte[bufferSize];
sdl.open(audioFmt, bufferSize);
sdl.start();
int numBytesRead = 0;
while((numBytesRead = ais.read(buffer)) != -1)
{
sdl.write(buffer, 0, numBytesRead);
}
}
catch (IOException | UnsupportedAudioFileException
| LineUnavailableException ex)
{
ex.printStackTrace();
}
}
private static void createAndShowGUI()
{
JFrame frame = new TriggerSound_SDL();
frame.setDefaultCloseOperation(DISPOSE_ON_CLOSE);
frame.pack();
frame.setVisible(true);
}
public static void main(String[] args)
{
SwingUtilities.invokeLater(() -> createAndShowGUI());
}
}
This code, with some small tweaks should let you at least test the different formats.
EDIT:
I'm seeing where your goal is to make a DAW!
In that case, you will want to convert the bytes to PCM data. Can I suggest you borrow some code from AudioCue? I basically wrote it to be a Clip-substitute, and part of that involved making the PCM data available for manipulation. Some techniques for mixing, playing back at different frequencies, multithreading can be found in it.

Thanks for all the advice guys. I will be getting rid of the ByteOutputStream and just use the AudioInputStream, I now understand what I was doing was unnecessary!! Thanks for the advice all! I have indeed tried using AudioCue but it is not low level enough for what I want to do!
One more thing guys. Previously, I created a multitrack media player which is using the Clip class. To play all the audio tracks together, I was looping through a list of Clips and playing them. However, this means that all tracks may be playing a tiny amount after each other due to the processing of the loop. Also, Clip class created a new thread per audio. I do not wants 100 threads running on 100 tracks, I want one thread for my audio output. I am still trying to work out how to start all tracks at the same time without a loop....(im guessing AudioCue have nailed the concurrent cues).
Does anyone know the best way to play multiple audio tracks in to one output? Do I need to route/bus all my audio tracks in to one output and somehow write all data from audio files in to one output buffer then play this output in a thread?
Thanks!!

how get available audio devices connected using java in mac osx

i was trying to get the available mixtures in MAC OSX using the below code. Even though i connected 3 different audio device, and able to see the same in system sound settings,below code doesn’t display all. i.e. mix.getMixerInfo()).isLineSupported(info) is not allowing to display anything and
even not able to connect using java .
The same is working fine with windows version.
public static void main(String[] args) {
String sf_ringtone = "/Users/abc.WAV";
AudioInputStream stream = null;
try {
stream = AudioSystem.getAudioInputStream(new File(sf_ringtone));
} catch (UnsupportedAudioFileException ex) {
System.out.println(ex.getStackTrace().toString());
} catch (IOException ex) {
System.out.println(ex.getStackTrace().toString());
}
AudioFormat format=null;
format = stream.getFormat();
if (format.getEncoding() != AudioFormat.Encoding.PCM_SIGNED) {
format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
format.getSampleRate(), format.getSampleSizeInBits() * 2,
format.getChannels(), format.getFrameSize() * 2,
format.getFrameRate(), true); // big endian
stream = AudioSystem.getAudioInputStream(format, stream);
}
DataLine.Info info = new DataLine.Info(Clip.class, stream.getFormat(),
((int) stream.getFrameLength() * format.getFrameSize()));
ArrayList<Mixer> mixerList = (ArrayList<Mixer>) getAllMixer();
for(Mixer mix:mixerList) {
System.out.println(" Mixture "+mix.getMixerInfo().getName());
}
ArrayList<String> cmbRingtonePlayback = new ArrayList<String>();
for (Mixer mix : mixerList) {
if (AudioSystem.getMixer(mix.getMixerInfo()).isLineSupported(info)) {
System.out.println(supported mixture :: "+mix.getMixerInfo().getName());
}
}
}
output:
Mixture Java Sound Audio Engine
Mixture Built-in Input
Mixture JABRA TALK
Mixture Logitech USB Headset
supported mixture :: Java Sound Audio Engine
Is this a limitation of sound API in mac osx ?, or is there any other way to do this in mac?

This issue faced was with apple Java for OS X 2014-001. Uninstalled java 1.6 from apple and tried with oracle java 1.7. Now this is working fine in both mac osx and windows.

Java AudioSystem: Reading 32 Bit Wav Files

I'm reading in audio files in 16 and 24 bit sampling bit depths and parsing them to determine their lengths without difficulty. However when reading a 32 bit file, I get
javax.sound.sampled.UnsupportedAudioFileException: could not get audio input stream from input file
at javax.sound.sampled.AudioSystem.getAudioInputStream(AudioSystem.java:1170)
...
The 32 bit test file is manually encoded in the same manner as the others (linear PCM). I'm wondering if AudioSystem doesn't support 32 bit Wavs, or if there might be a workaround. For reference, here's my class:
import java.io.*;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
public class soundUtility {
public static double getWavDuration(File filename)
{
AudioInputStream stream = null;
try
{
stream = AudioSystem.getAudioInputStream(filename);
AudioFormat format = stream.getFormat();
return filename.length() / format.getSampleRate() / (format.getSampleSizeInBits() / 8.0) / format.getChannels();
}
catch (Exception e)
{
e.printStackTrace();
return -1;
}
finally
{
try { stream.close(); } catch (Exception ex) { }
}
}
public static void main(String[] args) {
try {
// ===== TESTS: toggle these calls to test the included files =====
// File soundFile = new File("16bit.mono.441k.30secs.wav");
// File soundFile = new File("24bit.48k.11secs.stereo.wav");
File soundFile = new File("32bit.Floating.Stereo.48k.wav");
// ===========
System.out.println(getWavDuration(soundFile));
} catch (Exception e) {
e.printStackTrace();
}
}
}
Thanks for any insight.

Old question, but it's something I was just looking into for myself. My tests can confirm that 16bit 48kHz PCM works, and 32bit doesn't.
However my tests also imply that 24 bit doesn't work:
No line matching interface Clip supporting format PCM_SIGNED 48000.0 Hz, 24 bit, stereo, 6 bytes/frame, little-endian is supported.
This is a .wav file created at 96Khz, 32bit and exported from Audacity where it was rendered as a 24bit 48Khz wav.
This would seem reflected in the doc: https://docs.oracle.com/javase/8/docs/technotes/guides/sound/
"Sound formats: 8-bit and 16-bit audio data, in mono and stereo, with sample rates from 8 kHz to 48 kHz"
So, no 32 bit floating available, I'm afraid, and I cannot reproduce your result that 24 bit floating works.

Concatenation of two WAV files failed

I have this simple code to concatenate two wav files. Its pretty simple and the code runs without any errors. But there is a problem with the output file. The output file generated does not play, and surprisingly its size is only 44 bytes whereas my input files "a.wav" & "b.wav" are both more than 500Kb in size.
Here is my code:
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 WavConcat {
public static void main(String[] args) {
String wFile1 = "./sounds/a.wav";
String wFile2 = "./sounds/b.wav";
try {
AudioInputStream clip1 = AudioSystem.getAudioInputStream(new File(wFile1));
AudioInputStream clip2 = AudioSystem.getAudioInputStream(new File(wFile2));
AudioInputStream appendedFiles =
new AudioInputStream(
new SequenceInputStream(clip1, clip2),
clip1.getFormat(),
clip1.getFrameLength() + clip2.getFrameLength());
AudioSystem.write(appendedFiles,
AudioFileFormat.Type.WAVE,new File("./sounds/ab.wav"));
} catch (Exception e) {
e.printStackTrace();
}
}
}

Try this kind of structure. This worked for me
List audioInputStreamList = new ArrayList();
String wFile1 = "./sounds/a.wav";
String wFile2 = "./sounds/b.wav";
AudioInputStream audioInputStream1 = AudioSystem.getAudioInputStream(new File(wFile1));
AudioInputStream audioInputStream2 = AudioSystem.getAudioInputStream(new File(wFile2));
audioInputStreamList.add(audioInputStream1);
audioInputStreamList.add(audioInputStream2);
AudioFormat audioFormat = audioInputStream1.getFormat(); // audioInputStream2 format also same
AudioInputStream udioInputStream = new SequenceAudioInputStream(audioFormat,audioInputStreamList);
AudioSystem.write(audioInputStream, AudioFileFormat.Type.WAVE,new File("./sounds/ab.wav"));
UPDATE
check for SequenceAudioInputStream

clip1.getFormat() returns-->
MPEG2L3 24000.0 Hz, unknown bits per sample, mono, unknown frame size, 41.666668 frames/second
clip2.getFormat() returns-->
MPEG2L3 24000.0 Hz, unknown bits per sample, mono, unknown frame size, 41.666668 frames/second
That is an odd format. I can imagine the 'unknown bits per sample' is causing a problem, but also the MPEG2L3, since JavaSound has no inbuilt encoder for MP3. It seems like they are not encoded properly. Try loading them in sound editing software and save them as a type of WAV or AU that Java Sound can understand 'out of the box'. Hopefully the editing software:
Can understand the broken MP3, and..
Will write a valid WAV or AU.
If you can convert them to 8 bit mono & 8KHz during the conversion, it might reduce the byte[] size by a factor of 6 to 1. 8KHz is considered good enough to understand speech, and for this use you need to serve the bytes of the combined sound out to the browser - so reducing it in size is crucial.