I want to convert an audio file into a byte array. I currently did it and I want to know if its works :
private static AudioFormat getFormat() {
float sampleRate = 44100;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed,
bigEndian);
}
public static byte[] listenSound(File f) {
AudioInputStream din = null;
AudioInputStream outDin = null;
PCM2PCMConversionProvider conversionProvider = new PCM2PCMConversionProvider();
try {
AudioInputStream in = AudioSystem.getAudioInputStream(f);
AudioFormat baseFormat = in.getFormat();
AudioFormat decodedFormat = new AudioFormat(
AudioFormat.Encoding.PCM_SIGNED,
baseFormat.getSampleRate(),
16,
baseFormat.getChannels(),
baseFormat.getChannels() * 2,
baseFormat.getSampleRate(),
false);
din = AudioSystem.getAudioInputStream(decodedFormat, in);
if (!conversionProvider.isConversionSupported(getFormat(), decodedFormat)) {
System.out.println("Conversion Not Supported.");
System.exit(-1);
}
outDin = conversionProvider.getAudioInputStream(getFormat(), din);
ByteArrayOutputStream out = new ByteArrayOutputStream();
int n = 0;
byte[] buffer = new byte[1024];
while (true) {
n++;
if (n > 1000)
break;
int count = 0;
count = outDin.read(buffer, 0, 1024);
if (count > 0) {
out.write(buffer, 0, count);
}
}
in.close();
din.close();
outDin.close();
out.flush();
out.close();
//byte[] b=out.toByteArray();
//for(int i=0; i<b.length; i++)
//System.out.println("b = "+b[i]);
return out.toByteArray();
} catch (Exception e) {
e.printStackTrace();
}
return null;
}
That byte array data is actually known as time domain i need to be sure if it works before transforming this data into frequency domain with Discrete Fourier.
Thanks for your help !
typically the actual number of bytes processed is returned from a call like
count = outDin.read(buffer, 0, 1024);
so in addition to your current hard break after processing 1000 chunks if the API does in fact return a byte count you should check for it :
int size_chunk = 1024
byte[] buffer = new byte[size_chunk];
boolean keep_streaming = true;
while (keep_streaming) {
n++;
if (n > 1000) { // troubleshooting ONLY remove later
keep_streaming = false;
}
int count = 0;
count = outDin.read(buffer, 0, size_chunk);
if (count > 0) {
out.write(buffer, 0, count);
}
if (count < size_chunk) { // input stream has been consumed
keep_streaming = false;
}
}
You did not supply a link to the API doc so I cannot confirm, however assuming outDin.read will output the number of bytes actually processed, the above code will correctly output only the bytes to match input and so will result in a smaller output if input is less than 1 meg of data (your original logic blindly generated a 1 meg output stopped only after seeing 1000 chunks ... it also assumes you intend to truncate input after 1 meg of data as per your lines
if (n > 1000) { // troubleshooting ONLY remove later
keep_streaming = false;
}
Related
I have a 200kb file and I have to read it in bytes, and then write part of this byteArray (from index 90000 to 165000) to another file. How can I accomplish this.
File file1 = new File(getExternalStorageDirectory(), "file1.raw"); //the 200kb file
try {
File.createTempFile("file2", "mp3", getCacheDir()); //empty file that needs to get filled with part of the byte array from file1
} catch (IOException ignored) {}
Use a RandomAccessFile in order to seek to the offset to start copying from.
For example:
final int bufSize = 1024; // the buffer size for copying
byte[] buf = new byte[bufSize];
final long start = 10000L; // start offset at source
final long end = 15000L; // end (non-inclusive) offset at source
try (
RandomAccessFile in = new RandomAccessFile("input.bin", "r");
OutputStream out = new FileOutputStream("output.bin"))
{
in.seek(start);
long remaining = end - start;
do {
int n = (remaining < bufSize)? ((int) remaining) : bufSize;
int nread = in.read(buf, 0, n);
if (nread < 0)
break; // EOF
remaining -= nread;
out.write(buf, 0, nread);
} while (remaining > 0);
}
I obtaining bytes from InputStream, but I need to modify and save them to Wav File.
Here my code:
Socket Sending Audio Obtained from Microphone.
AudioFormat adfmt = new AudioFormat(8000.0f, 8, 1, true , true);
int bufferSize = (int) adfmt.getSampleRate()* adfmt.getFrameSize();
byte[] buffer = new byte[bufferSize];
Socket clientSocketO = new Socket(...);
OutputStream output = clientSocketO.getOutputStream();
DataLine.Info dlInfo = new DataLine.Info(TargetDataLine.class, adfmt);
TargetDataLine tdLine = (TargetDataLine) AudioSystem.getLine(dlInfo);
tdLine.open(adfmt);
tdLine.start(); // start capturing
boolean bRunningO = true;
while (bRunningO) {
int count = tdLine.read(buffer, 0, buffer.length);
if (count > 0) {
byte[] outgoingBytes = Arrays.copyOf(buffer, count);
output.write(outgoingBytes);
}
}
tdLine.flush();
In the Other Side Socket receiving bytes :
AudioFormat adfmt = new AudioFormat(8000.0f, 8, 1, true , true);
int bufferSize = (int) adfmt.getSampleRate()* adfmt.getFrameSize();
byte[] buffer = new byte[bufferSize];
Socket clientSocketI = new Socket(...);
InputStream input = clientSocketI.getInputStream();
String fileName = System.getProperty("file.separator") + "SomeFile.wav"
File fileStreamedWav = new File((new File("")).getAbsolutePath() + fileName);
AudioInputStream ais;
ByteArrayInputStream bis;
DataLine.Info dlInfo = new DataLine.Info(SourceDataLine.class, adfmt);
//SourceDataLine sdLine = (SourceDataLine) AudioSystem.getLine(dlInfo);
//sdLine.open(adfmt);
//sdLine.start(); // start playback
AudioFileFormat.Type afType = AudioFileFormat.Type.WAVE;
boolean bRunningI = true;
while (bRunningI) {
try {
int read = input.read(buffer); //Socket Reading bytes
byte[] incomingBytes;
if (read > 0) {
incomingBytes = Arrays.copyOf(buffer, read);
if (incomingBytes!= null) {
//sdLine.write(incomingBytes, 0, incomingBytes.length);
//Same Size bytes, but isn't necessary submit the put Code
byte[] changedBytes = MethodChangerBytes(incomingBytes);
bis = new ByteArrayInputStream(changedBytes);
ais = new AudioInputStream(bis, adfmt,
changedBytes.length/adfmt.getFrameSize());
int W = AudioSystem.write(ais, afType, fileStreamedWav);
System.out.println("AudioSystem.write:" + W);
}
}
} catch (IOException e) {
bRunningI = false;
}
}
Here the code modifier of Bytes, for Now assume amplify by two...
byte[] MethodChangerBytes(byte[] incoming) {
byte[] outgoing = new byte[incoming.length];
for (int i = 0; i < incoming.length; i ++) {
// Really is not important what happens here
double Sample = (double)(short)(((incoming[i] - 128) & 0xFF) << 8);
Sample *= 2.0;
outgoing[i] = (byte)(((int()Sample >> 8) + 128) & 0xFF);
}
return outgoing;
}
When sdLine is uncommented then I can here all sound transmitted.
AudioInputStream(InputStream stream, AudioFormat format, long length)
AudioSystem.write(AudioInputStream stream, AudioFileFormat.Type fileType, File out)
The problem:
This code Only Save the Last Bytes obtained from MethodChangerBytes.
Question:
How Save all bytes processed Wav bytes until Socket connection is closed?
Thank you
Have a buffer:
ByteArrayOutputStream outputStream=new ByteArrayOutputStream();
write to this buffer then move the writing outside the loop; when all bytes are read save:
boolean bRunningI = true;
try {
while (bRunningI) {
int read = input.read(buffer); //Socket Reading bytes
byte[] incomingBytes;
if (read > 0) {
incomingBytes = Arrays.copyOf(buffer, read);
if (incomingBytes!= null) {
//sdLine.write(incomingBytes, 0, incomingBytes.length);
//Same Size bytes, but isn't necessary submit the put Code
byte[] changedBytes = MethodChangerBytes(incomingBytes);
outputStream.write(changedBytes, 0, changedBytes.length);
}
}
}
byte[] allBytes=outputStream.toByteArray();
bis = new ByteArrayInputStream(allBytes);
ais = new AudioInputStream(bis, adfmt,
changedBytes.length/adfmt.getFrameSize());
int W = AudioSystem.write(ais, afType, fileStreamedWav);
System.out.println("AudioSystem.write:" + W);
} catch (IOException e) {
bRunningI = false;
}
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!!!
}
}
First of all, if not using function decode_path , I can play .wav file with my code , and it works fine I use Jlayer and audio track to play the song.
Second, if I use function decode_path it can decode mp3 to pcm file , and pass the byte[] to function PlayAudioTrack, and let it play.
The quesion is,I don't know where my code is wrong , I use 320Kbps, 44.1Khz stereo type, Layer3 mp3, but the AudioTrack plays noise but no music~!!!!
can anyone ?
???
My code
public void PlayAudioTrack(String filePath) throws IOException{
int intSize = android.media.AudioTrack.getMinBufferSize(44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT);
AudioTrack at = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT, intSize, AudioTrack.MODE_STREAM);
//Reading the file..
int count = 512 * 1024; // 512 kb
// byte[] byteData = null;
// byteData = new byte[(int)count];
//we can decode correct byte data here
byte[] byteData = null;
byteData = decode_path(filePath, 0, 20000);
File file = null;
file = new File(filePath);
FileInputStream in = null;
try {
in = new FileInputStream( file );
} catch (FileNotFoundException e) {
e.printStackTrace();
}
int bytesread = 0, ret = 0;
int size = (int) file.length();
at.play();
while (bytesread < size) {
Log.e("devon","write byte array with sizes");
ret = in.read( byteData,0, count);
if (ret != -1) {
Log.e("devon","Write the byte array to the track");
at.write(byteData,0, ret);
bytesread += ret;
}else break;
}
at.stop();
at.release();
}
public static byte[] decode_path(String path, int startMs, int maxMs)
throws IOException{
ByteArrayOutputStream outStream = new ByteArrayOutputStream(1024);
float totalMs = 0;
boolean seeking = true;
File file = new File(path);
InputStream inputStream = new BufferedInputStream(new FileInputStream(file), 8 * 1024);
try {
Bitstream bitstream = new Bitstream(inputStream);
Decoder decoder = new Decoder();
boolean done = false;
while (! done) {
Header frameHeader = bitstream.readFrame();
if (frameHeader == null) {
done = true;
} else {
totalMs += frameHeader.ms_per_frame();
if (totalMs >= startMs) {
seeking = false;
}
if (! seeking) {
SampleBuffer output = (SampleBuffer) decoder.decodeFrame(frameHeader, bitstream);
if (output.getSampleFrequency() != 44100
|| output.getChannelCount() != 2) {
throw new IllegalArgumentException("mono or non-44100 MP3 not supported");
}
short[] pcm = output.getBuffer();
for (short s : pcm) {
outStream.write(s & 0xff);
outStream.write((s >> 8 ) & 0xff);
}
}
if (totalMs >= (startMs + maxMs)) {
done = true;
}
}
bitstream.closeFrame();
}
return outStream.toByteArray();
} catch (BitstreamException e) {
throw new IOException("Bitstream error: " + e);
} catch (DecoderException e) {
Log.w(TAG, "Decoder error", e);
throw new IOException("Decoder error: " + e);
}
}
public void PlayAudioTrack(String filePath) throws IOException{
int intSize = android.media.AudioTrack.getMinBufferSize(44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT);
AudioTrack at = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT, intSize, AudioTrack.MODE_STREAM);
//Reading the file..
int count = 512 * 1024; // 512 kb
// byte[] byteData = null;
// byteData = new byte[(int)count];
//we can decode correct byte data here
byte[] byteData = null;
byteData = decode_path(filePath, 0, 20000);
int temp =0;
at.play();
while (temp<byteData.length)
{
at.write(byteData, temp, count);
temp+= count;
}
at.stop();
at.release();
}
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!!!
}
}