Oh you wish to find the volume? Well, not all hardware supports it, but here is how you get the dataline.
public static SourceDataLine getSourceDataLine(Line.Info lineInfo){
try{
return (SourceDataLine) AudioSystem.getLine(lineInfo);
}
catch(Exception ex){
ex.printStackTrace();
return null;
}
}
Then just call SourceDataLine.getLevel() to get the volume. I hope this helps.
NB: If the sound is originating from outside the JVM or not via the JavaSound API, this method will not detect the sound as the JVM does not have access to the OS equivalent of the SourceDataLine.
UPDATE: Upon further research, getLevel() is not implemented on most Systems. So I have manually implemented the method based off this forum discussion: https://community.oracle.com/message/5391003
Here are the classes:
public class Main {
public static void main(String[] args){
MicrophoneAnalyzer mic = new MicrophoneAnalyzer(FLACFileWriter.FLAC);
System.out.println("HELLO");
mic.open();
while(true){
byte[] buffer = new byte[mic.getTargetDataLine().getFormat().getFrameSize()];
mic.getTargetDataLine().read(buffer, 0, buffer.length);
try{
System.out.println(getLevel(mic.getAudioFormat(), buffer));
}
catch(Exception e){
System.out.println("ERROR");
e.printStackTrace();
}
}
}
public static double getLevel(AudioFormat af, byte[] chunk) throws IOException{
PCMSigned8Bit converter = new PCMSigned8Bit(af);
if(chunk.length != converter.getRequiredChunkByteSize())
return -1;
AudioInputStream ais = converter.convert(chunk);
ais.read(chunk, 0, chunk.length);
long lSum = 0;
for(int i=0; i<chunk.length; i++)
lSum = lSum + chunk[i];
double dAvg = lSum / chunk.length;
double sumMeanSquare = 0d;
for(int j=0; j<chunk.length; j++)
sumMeanSquare = sumMeanSquare + Math.pow(chunk[j] - dAvg, 2d);
double averageMeanSquare = sumMeanSquare / chunk.length;
return (Math.pow(averageMeanSquare,0.5d));
}
}
The method I used only works on 8bitPCM so we have to convert the encoding to that using these two classes. Here is the general abstract converter class.
import java.io.ByteArrayInputStream;
import java.io.IOException;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
abstract class AbstractSignedLevelConverter
{
private AudioFormat srcf;
public AbstractSignedLevelConverter(AudioFormat sourceFormat)
{
srcf = sourceFormat;
}
protected AudioInputStream convert(byte[] chunk)
{
AudioInputStream ais = null;
if(AudioSystem.isConversionSupported( AudioFormat.Encoding.PCM_SIGNED,
srcf))
{
if(srcf.getEncoding() != AudioFormat.Encoding.PCM_SIGNED)
ais = AudioSystem.getAudioInputStream(
AudioFormat.Encoding.PCM_SIGNED,
new AudioInputStream(new ByteArrayInputStream(chunk),
srcf,
chunk.length * srcf.getFrameSize()));
else
ais = new AudioInputStream(new ByteArrayInputStream(chunk),
srcf,
chunk.length * srcf.getFrameSize());
}
return ais;
}
abstract public double convertToLevel(byte[] chunk) throws IOException;
public int getRequiredChunkByteSize()
{
return srcf.getFrameSize();
}
}
And here is the one for 8BitPCM
import java.io.IOException;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
public class PCMSigned8Bit extends AbstractSignedLevelConverter
{
PCMSigned8Bit(AudioFormat sourceFormat)
{
super(sourceFormat);
}
public double convertToLevel(byte[] chunk) throws IOException
{
if(chunk.length != getRequiredChunkByteSize())
return -1;
AudioInputStream ais = convert(chunk);
ais.read(chunk, 0, chunk.length);
return (double)chunk[0];
}
}
This is for TargetDataLine which may not work in your use case, but you could build a wrapper around SourceDataLine and use this to properly implement these methods. Hopes this helps.