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Java: Processing audio in real time from speakers

I'm working on a project that will change the brightness of keyboard lights with PWM signal, according to output sound levels.
That trouble is, I need the output stream in real time. So far I used javax.sound.sampled package and I never succeeded grabbing the audio output.
However, what have I done was to use targetDataLine and sourceDataLine, which don't look like the way to go, I'm still trying though.
What I need is an audio stream provider, which my java app can "hear" and process it.
So far, I viewed many tutorials, videos, posts, some articles (like this one: https://cr.openjdk.java.net/~iris/se/12/latestSpec/api/java.desktop/javax/sound/sampled/class-use/Mixer.Info.html ) etc. but no results.
Has somebody done it before ? Or is out there a library other than sampled library ?
Any help would be appreciated.
The error I get for every single format:
java.lang.IllegalArgumentException: Line unsupported: interface TargetDataLine supporting format PCM_SIGNED 44100.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian
at com.sun.media.sound.DirectAudioDevice.getLine(Unknown Source)
at javax.sound.sampled.AudioSystem.getTargetDataLine(Unknown Source)
The code used:
package audio;
import javax.sound.sampled.*;
public class App1 {
public static String getWord(String arr) {
int i = 0;
for(; i < arr.length(); i++) {
if (arr.charAt(i) == ' ')
break;
}
return arr.substring(0,i);
}
public static void wait(int ms) {
try{
Thread.sleep(ms);
}
catch(InterruptedException ex){
Thread.currentThread().interrupt();
}
}
public static AudioFormat getAudioFormat(){
float sampleRate = 44100;
//8000,11025,16000,22050,44100
int sampleSizeInBits = 16;
//8,16
int channels = 1;
//1,2
boolean signed = true;
//true,false
boolean bigEndian = false;
//true,false
//return new AudioFormat(Encoding.PCM_SIGNED, sampleRate, 16, 1, 2, sampleRate, false);
return new AudioFormat(sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}
public static void main(String [] args) {
try {
//-----DECLARATIONS-----
TargetDataLine targetDataLine = null;
Mixer.Info[] mixers = AudioSystem.getMixerInfo();
Mixer.Info m = null;
String expectedMixer = "Speakers";
//-----MIXER FINDER-----
System.out.println("Number of mixers: " + mixers.length);
for(int i = 0; i < mixers.length; i++) {
System.out.println(getWord(mixers[i].getName()));
if(getWord(mixers[i].getName()).compareTo(expectedMixer) == 0){
m = mixers[i];
}
}
if(m==null) throw new Exception("No such mixer found: " + expectedMixer);
else System.out.println('\n'+"Device choosen: "+m.getName());
//-----LINE TESTER-----
boolean v = false, showError = true; // show error or keep trying several times
int tries = 3, i = 0;
while(v==false && i++ < tries){
try {
//sourceDataLine = AudioSystem.getTargetDataLine(getAudioFormat(), m);
targetDataLine = AudioSystem.getTargetDataLine(getAudioFormat(), m);
targetDataLine.open(getAudioFormat());
v=true;
//System.out.println("Success!");
} catch (IllegalArgumentException e){
if (showError) {
v = true;
e.printStackTrace();
}
else {
System.out.println("Error! Retrying... "+i+'/'+tries);
v = false;
}
wait(2000);
}
}
if(i-1==tries)
//System.out.println("No success :(");
throw new Exception("No success :(");
else
if(v==false)
System.out.println("SourceData line found and accepted !");
//-----SIGNAL PROCESSING-----
//nothing here because the rest isn't working
} catch(Exception e) { e.printStackTrace();}
}
}
Lately, I've been searching for some piece of code to discover available formats, and I've made something between my code and edoloughlin's code:
package audio;
import javax.sound.sampled.*;
public class App2 {
public static String getWord(String arr) {
int i = 0;
for(; i < arr.length(); i++) {
if (arr.charAt(i) == ' ')
break;
}
return arr.substring(0,i);
}
public static void wait(int ms) {
try{
Thread.sleep(ms);
}
catch(InterruptedException ex){
Thread.currentThread().interrupt();
}
}
public static AudioFormat getAudioFormat(){
float sampleRate = 44100;
//8000,11025,16000,22050,44100
int sampleSizeInBits = 16;
//8,16
int channels = 1;
//1,2
boolean signed = true;
//true,false
boolean bigEndian = false;
//true,false
//return new AudioFormat(Encoding.PCM_SIGNED, sampleRate, 16, 1, 2, sampleRate, false);
return new AudioFormat(sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}
public static void main(String [] args) {
try {
String expectedMixer = "Speakers";
Mixer.Info[] mixers = AudioSystem.getMixerInfo();
Mixer.Info m = null;
for(int i = 0; i < mixers.length; i++) {
if(getWord(mixers[i].getName()).compareTo(expectedMixer) == 0){
m = mixers[i];
}
}
int sampleRates[] = { 8000, 11025, 16000, 22050, 44100 };
int channels[] = { 1, 2 };
int bytesPerSample[] = { 1, 2 };
boolean signature[] = {true, false};
AudioFormat format;
DataLine.Info lineInfo;
//for (Mixer.Info mixerInfo : AudioSystem.getMixerInfo()) {
for (int a = 0; a < sampleRates.length; a++) {
for (int b = 0; b < channels.length; b++) {
for (int c = 0; c < bytesPerSample.length; c++) {
for(int d = 0; d < signature.length; d++) {
format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
(float)sampleRates[a], 8 * bytesPerSample[c], channels[b], bytesPerSample[c],
(float)sampleRates[a], signature[d]);
lineInfo = new DataLine.Info(/*dataLineClass*/TargetDataLine.class, format);
if (AudioSystem.isLineSupported(lineInfo)) {
/*
* TODO: To perform an exhaustive search on supported lines, we should open
* TODO: each Mixer and get the supported lines. Do this if this approach
* TODO: doesn't give decent results. For the moment, we just work with whatever
* TODO: the unopened mixers tell us.
*/
if (AudioSystem.getMixer(/*mixerInfo*/m).isLineSupported(lineInfo)) {
//formats.add(format);
System.out.println(format);
}
}
}
}
}
}
//}
}catch(Exception e) {e.printStackTrace();}
}
}
As shown above, I used many format combinations, but nothing gets printed out in the console. I wonder if my system/java app supports such a task. If not, is there a way to achieve the goal ? (read real-time output)

MediaRecorder record audio in a loop

I'm developing a sound recognition system. I'm using a tensorflow model developed on python to convert MFCC values to labels. I'm using the MediaRecorder class to record the audio, and I'm doing it in a loop so I can be constantly getting microphone audio and then getting the label from the model. Here is the recording loop:
temp = 0;
while (true) {
audioPath = getApplicationContext().getFilesDir().getAbsolutePath();
audioPath += "/Recording" + temp + ".3gp";
audioFile = new File(audioPath);
mediaRecorder = new MediaRecorder();
mediaRecorder.setAudioSource(MediaRecorder.AudioSource.MIC);
mediaRecorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP);
mediaRecorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
mediaRecorder.setOutputFile(audioPath);
try {
mediaRecorder.prepare();
} catch (IOException e) {
e.printStackTrace();
}
mediaRecorder.start();
sleep(2000);
if (!isRunning) {
mediaRecorder.stop();
return;
}
try {
int amplitude = mediaRecorder.getMaxAmplitude();
Log.d("volume", Integer.toString(amplitude));
//finished = false;
avgVolumeTask task = new avgVolumeTask();
task.execute(amplitude);
} catch (Exception e) {
Log.d("Exception in startMediaRecorder()", e.toString());
}
mediaRecorder.stop();
mediaRecorder.release();
soundRecognition task2 = new soundRecognition();
task2.execute();
audioFile.delete();
temp++;
}
This is the soundRecognition method:
private class soundRecognition extends AsyncTask<Integer, Integer, Long> {
#Override
protected Long doInBackground(Integer... level) {
float[] mfccValues = null;
Interpreter tflite = null;
float[][] labelProbArray = null;
try {
mfccValues = computeMFCC();
labelList = loadLabelList();
labelProbArray = new float[1][labelList.size()];
tflite = new Interpreter(loadModel());
} catch (IOException e) {
e.printStackTrace();
} catch (UnsupportedAudioFileException e) {
e.printStackTrace();
}
tflite.run(mfccValues, labelProbArray);
for (int i = 0; i < labelProbArray[0].length; i++) {
float value = labelProbArray[0][i];
//if (i == 1f){
//Log.d("Output at " + Integer.toString(i) + ": ", Float.toString(value));
//doAlert(i);
//}
}
return null;
}
}
The computeMFCC method is this:
public float[] computeMFCC() throws IOException, UnsupportedAudioFileException {
FileInputStream in2 = new FileInputStream(audioPath);
int i;
// InputStream to byte array
byte[] buf = IOUtils.toByteArray(in2);
in2.close();
i = Integer.MAX_VALUE;
// byte array to short array
short[] shortArr = new short[buf.length / 2];
ByteBuffer.wrap(buf).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shortArr);
int count = 0;
while (count <= shortArr.length) { // Still have data to process.
for (int n = 0; n < nSubframePerBuf; n++) { // Process audio signal in ArrayList and shift by one subframe each time
int k = 0;
for (i = (n * frameShift); i < (n + 1) * frameShift; i++) {
subx[k] = shortArr[i];
k++;
}
subframeList.add(subx); // Add the current subframe to the subframe list. Later, a number of
}
count++;
}
// Need at least nSubframePerMfccFrame to get one analysis frame
x = extractOneFrameFromList(nSubframePerMfccFrame);
MFCC mfcc = new MFCC(samplePerFrm, 16000, numMfcc);
double[] mfccVals = mfcc.doMFCC(x);
float[] floatArray = new float[mfccVals.length];
for (i = 0 ; i < mfccVals.length; i++)
{
floatArray[i] = (float) mfccVals[i];
}
return floatArray;
}
And the doMFCC method is from a downloaded java file here:
https://github.com/enmwmak/ScreamDetector/blob/master/src/edu/polyu/mfcc/MFCC.java
The issue I'm having is that after a few iterations, I run into the problem that the file doesnt get created, and then get a null error passing the results from the input stream to the tensorflow model.
Possible Issues
One reason could be where the file is stored. I've been trying to send the file to local storage because I was worried that all the devices wouldnt have external storage.
Another reason could be that i'm not calling the sound recognition in the right spot. I waited will after the mediaRecorder is stopped to make sure that the file is written with the mic audio, but when I review the contents of the fileInputStream, it appears to not be working, and in each loop the file is always the same.
Any help would be much appreciated.

It may be tricky to have a sleep(2000) inside while loop.
It may be better to check millis and break until 2000 ms has lapsed.

Writing to a file using a byte stream

I'm trying to write a series of 10000 random integers to a text file using a byte stream, however once I open the text file up it has a collection of random characters which seemingly have little to do with the integer values I want to be shown. I'm new to this form of stream, I'm guessing that the integer values are being taken as byte values, but I can't think of a way to get round this.
My current attempt...
import java.io.FileOutputStream;
import java.io.IOException;
import java.util.Random;
public class Question1ByteStream {
public static void main(String[] args) throws IOException {
FileOutputStream out = new FileOutputStream("ByteStream.txt");
try {
for(int i = 0; i < 10000; i ++){
Integer randomNumber = randInt(0, 100000);
int by = randomNumber.byteValue();
out.write(by);
}
}finally{
if (out != null) {
out.close();
}
}
}
public static int randInt(int min, int max) {
Random rand = new Random();
int randomNum = rand.nextInt((max - min) + 1) + min;
return randomNum;
}
}
Apologies if this lacks clarity.

It's because the numbers that you write are not written as strings into the txt but as raw byte value.
Try the following code:
BufferedWriter writer = null;
try {
writer = new BufferedWriter(new FileWriter("./output.txt"));
writer.write(yourRandomNumberOfTypeInteger.toString());
} catch (IOException e) {
System.err.println(e);
} finally {
if (writer != null) {
try {
writer.close();
} catch (IOException e) {
System.err.println(e);
}
}
}
Or, if referring to your original code, write the Integer directly:
try {
for(int i = 0; i < 10000; i ++){
Integer randomNumber = randInt(0, 100000);
out.write(randomNumber.toString());
}
}finally{
if (out != null) {
out.close();
}
}

dont do like below(writing in the form of byte characters)
for(int i = 0; i < 10000; i ++){
Integer randomNumber = randInt(0, 100000);
int by = randomNumber.byteValue();
out.write(by);
}
write it in the form of string as it is a text file
for(int i = 0; i < 10000; i ++){
Integer randomNumber = randInt(0, 100000);
out.write(randomNumber);
}
automatically toString() method will be called for Integer Object randomNumber
and it will be written to file.

Split Wave audio file at silence [duplicate]

How can I detect silence when recording operation is started in Java? What is PCM data? How can I calculate PCM data in Java?
I found the solution :
package bemukan.voiceRecognition.speechToText;
import javax.sound.sampled.*;
import java.io.*;
public class RecordAudio {
private File audioFile;
protected boolean running;
private ByteArrayOutputStream out;
private AudioInputStream inputStream;
final static float MAX_8_BITS_SIGNED = Byte.MAX_VALUE;
final static float MAX_8_BITS_UNSIGNED = 0xff;
final static float MAX_16_BITS_SIGNED = Short.MAX_VALUE;
final static float MAX_16_BITS_UNSIGNED = 0xffff;
private AudioFormat format;
private float level;
private int frameSize;
public RecordAudio(){
getFormat();
}
private AudioFormat getFormat() {
File file = new File("src/Facebook/1.wav");
AudioInputStream stream;
try {
stream = AudioSystem.getAudioInputStream(file);
format=stream.getFormat();
frameSize=stream.getFormat().getFrameSize();
return stream.getFormat();
} catch (UnsupportedAudioFileException e) {
} catch (IOException e) {
}
return null;
}
public void stopAudio() {
running = false;
}
public void recordAudio() {
try {
final AudioFormat format = getFormat();
DataLine.Info info = new DataLine.Info(
TargetDataLine.class, format);
final TargetDataLine line = (TargetDataLine)
AudioSystem.getLine(info);
line.open(format);
line.start();
Runnable runner = new Runnable() {
int bufferSize = (int) format.getSampleRate()
* format.getFrameSize();
byte buffer[] = new byte[bufferSize];
public void run() {
int readPoint = 0;
out = new ByteArrayOutputStream();
running = true;
int sum=0;
while (running) {
int count =
line.read(buffer, 0, buffer.length);
calculateLevel(buffer,0,0);
System.out.println(level);
if (count > 0) {
out.write(buffer, 0, count);
}
}
line.stop();
}
};
Thread captureThread = new Thread(runner);
captureThread.start();
} catch (LineUnavailableException e) {
System.err.println("Line unavailable: " + e);
System.exit(-2);
}
}
public File getAudioFile() {
byte[] audio = out.toByteArray();
InputStream input = new ByteArrayInputStream(audio);
try {
final AudioFormat format = getFormat();
final AudioInputStream ais =
new AudioInputStream(input, format,
audio.length / format.getFrameSize());
AudioSystem.write(ais, AudioFileFormat.Type.WAVE, new File("temp.wav"));
input.close();
System.out.println("New file created!");
} catch (IOException e) {
System.out.println(e.getMessage());
}
return new File("temp.wav");
}
private void calculateLevel (byte[] buffer,
int readPoint,
int leftOver) {
int max = 0;
boolean use16Bit = (format.getSampleSizeInBits() == 16);
boolean signed = (format.getEncoding() ==
AudioFormat.Encoding.PCM_SIGNED);
boolean bigEndian = (format.isBigEndian());
if (use16Bit) {
for (int i=readPoint; i<buffer.length-leftOver; i+=2) {
int value = 0;
// deal with endianness
int hiByte = (bigEndian ? buffer[i] : buffer[i+1]);
int loByte = (bigEndian ? buffer[i+1] : buffer [i]);
if (signed) {
short shortVal = (short) hiByte;
shortVal = (short) ((shortVal << 8) | (byte) loByte);
value = shortVal;
} else {
value = (hiByte << 8) | loByte;
}
max = Math.max(max, value);
} // for
} else {
// 8 bit - no endianness issues, just sign
for (int i=readPoint; i<buffer.length-leftOver; i++) {
int value = 0;
if (signed) {
value = buffer [i];
} else {
short shortVal = 0;
shortVal = (short) (shortVal | buffer [i]);
value = shortVal;
}
max = Math.max (max, value);
} // for
} // 8 bit
// express max as float of 0.0 to 1.0 of max value
// of 8 or 16 bits (signed or unsigned)
if (signed) {
if (use16Bit) { level = (float) max / MAX_16_BITS_SIGNED; }
else { level = (float) max / MAX_8_BITS_SIGNED; }
} else {
if (use16Bit) { level = (float) max / MAX_16_BITS_UNSIGNED; }
else { level = (float) max / MAX_8_BITS_UNSIGNED; }
}
} // calculateLevel
}

How can I detect silence when recording operation is started in Java?
Calculate the dB or RMS value for a group of sound frames and decide at what level it is considered to be 'silence'.
What is PCM data?
Data that is in Pulse-code modulation format.
How can I calculate PCM data in Java?
I do not understand that question. But guessing it has something to do with the speech-recognition tag, I have some bad news. This might theoretically be done using the Java Speech API. But there are apparently no 'speech to text' implementations available for the API (only 'text to speech').
I have to calculate rms for speech-recognition project. But I do not know how can I calculate in Java.
For a single channel that is represented by signal sizes in a double ranging from -1 to 1, you might use this method.
/** Computes the RMS volume of a group of signal sizes ranging from -1 to 1. */
public double volumeRMS(double[] raw) {
double sum = 0d;
if (raw.length==0) {
return sum;
} else {
for (int ii=0; ii<raw.length; ii++) {
sum += raw[ii];
}
}
double average = sum/raw.length;
double sumMeanSquare = 0d;
for (int ii=0; ii<raw.length; ii++) {
sumMeanSquare += Math.pow(raw[ii]-average,2d);
}
double averageMeanSquare = sumMeanSquare/raw.length;
double rootMeanSquare = Math.sqrt(averageMeanSquare);
return rootMeanSquare;
}
There is a byte buffer to save input values from the line, and what I should have to do with this buffer?
If using the volumeRMS(double[]) method, convert the byte values to an array of double values ranging from -1 to 1. ;)

You need to catch the value like a number silence is zero or near
Please adapt your code to your requirement!!!
In this case a variable named UMBRAL (Threshold in spanish)...
Suppose that you have access to WAV file like bytes ByteHeader...
private Integer Byte2PosIntBig(byte Byte24, byte Byte16, byte Byte08, byte Byte00) {
return new Integer (
((Byte24) << 24)|
((Byte16 & 0xFF) << 16)|
((Byte08 & 0xFF) << 8)|
((Byte00 & 0xFF) << 0));
}
Before ....
RandomAccessFile RAFSource = new RandomAccessFile("your old file wav", "r");
Begins here...
int PSData = 44;
byte[] Bytes = new byte[4];
byte[] ByteHeader = new byte[44];
RAFSource.seek(0);
RAFSource.read(ByteHeader);
int WavSize = Byte2PosIntBig(ByteHeader[43],ByteHeader[42],ByteHeader[41],ByteHeader[40]);
int NumBits = Byte2PosIntBig(ByteHeader[35],ByteHeader[34]);
int NumByte = NumBits/8;
for (int i = PSData;i < PSData+WavSize;i+=NumByte) {
int WavSample = 0;
int WavResultI =0;
int WavResultO = 0;
if (NumByte == 2) {
RAFSource.seek(i);
Bytes[0] = RAFSource.readByte();
Bytes[1] = RAFSource.readByte();
WavSample = (int)(((Bytes[1]) << 8)|((Bytes[0] & 0xFF) << 0));
if (Math.abs(WavSample) < UMBRAL) {
//SILENCE DETECTED!!!
}
} else {
RAFSource.seek(i);
WavSample = (short)(RAFSource.readByte() & 0xFF);
short sSamT = (short)WavSample;
sSamT += 128;
double dSamD = (double)sSamT*Multiplier;
if ((double)sSamT < UMBRAL) {
//SILENCE DETECTED!!!
}
}

Extracting Big Wav file into smaller chunks in Java

I have a big wav file that I would like to get into smaller chunks. I also have a .cue file that have the frame rate lengths, at which the smaller chunks should be. I figured out how to split the wav up, but all the wav files that are made are the same sound. It seems that everytime I create a new wav the big wav file is starting from the beginning and making the new wave the correct length but same sound.
I think I need a way to read the wav to a specific frame, then write to a file, then continue reading and write to another file,etc...
I've been at this for hours and can't seem to figure it out. any help would be greatly appreciated. Here is my code, all the commented stuff is my wrong code that I have been trying.
int count2 = 0;
int totalFramesRead = 0;
//cap contains the how many wav's are to be made
//counter contains the vector position.
String wavFile1 = "C:\\Users\\DC3\\Desktop\\wav&text\\testwav.wav";
//String wavFile2 = "C:\\Users\\DC3\\Desktop\\waver\\Battlefield.wav";
while(count2 != counter){
try {
AudioInputStream clip1 = AudioSystem.getAudioInputStream(new File(wavFile1));
int bytesPerFrame = clip1.getFormat().getFrameSize();
//System.out.println(bytesPerFrame);
// int numBytes = safeLongToInt(clip1.getFrameLength()) * bytesPerFrame;
// byte[] audioBytes = new byte[numBytes];
// int numBytesRead = 0;
// int numFramesRead = 0;
// // Try to read numBytes bytes from the file.
// while ((numBytesRead =
// clip1.read(audioBytes)) != -1) {
// // Calculate the number of frames actually read.
// clip1.read(audioBytes)
// numFramesRead = numBytesRead / bytesPerFrame;
// totalFramesRead += numFramesRead;
// System.out.println(totalFramesRead);
// }
long lengthofclip = Integer.parseInt(time.get(count2))- silence;
globallength = clip1.getFrameLength();
AudioInputStream appendedFiles = new AudioInputStream(clip1, clip1.getFormat(), lengthofclip);
//long test = (appendedFiles.getFrameLength() *24 *2)/8;
//int aaaaa = safeLongToInt(test);
//appendedFiles.mark(aaaaa);
AudioSystem.write(appendedFiles,
AudioFileFormat.Type.WAVE,
new File("C:\\Users\\DC3\\Desktop\\wav&text\\" + name.get(count2)));
count2++;
} catch (Exception e) {
e.printStackTrace();
}
}
}
public static int safeLongToInt(long l) {
if (l < Integer.MIN_VALUE || l > Integer.MAX_VALUE) {
throw new IllegalArgumentException
(l + " cannot be cast to int without changing its value.");
}
return (int) l;
}

Just a thought at first glance but I'm assuming it's this line giving trouble:
AudioInputStream clip1 = AudioSystem.getAudioInputStream(new File(wavFile1));
Take that and put it outside of your while loop so it doesn't get recreated every cycle. Like so:
//...
String wavFile1 = "C:\\Users\\DC3\\Desktop\\wav&text\\testwav.wav";
AudioInputStream clip1 = AudioSystem.getAudioInputStream(new File(wavFile1));
int bytesPerFrame = clip1.getFormat().getFrameSize();
while(count2 != counter){
try {
//...
This also assumes that your algorithm is correct, which I'm not going to waste time thinking about because you didn't ask that question :-D