/* A buffer supporting the capturing of audio samples from a file or
from the computer's audio input port.
Copyright (c) 2000-2006 The Regents of the University of California.
All rights reserved.
Permission is hereby granted, without written agreement and without
license or royalty fees, to use, copy, modify, and distribute this
software and its documentation for any purpose, provided that the above
copyright notice and the following two paragraphs appear in all copies
of this software.
IN NO EVENT SHALL THE UNIVERSITY OF CALIFORNIA BE LIABLE TO ANY PARTY
FOR DIRECT, INDIRECT, SPECIAL, INCIDENTAL, OR CONSEQUENTIAL DAMAGES
ARISING OUT OF THE USE OF THIS SOFTWARE AND ITS DOCUMENTATION, EVEN IF
THE UNIVERSITY OF CALIFORNIA HAS BEEN ADVISED OF THE POSSIBILITY OF
SUCH DAMAGE.
THE UNIVERSITY OF CALIFORNIA SPECIFICALLY DISCLAIMS ANY WARRANTIES,
INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE SOFTWARE
PROVIDED HEREUNDER IS ON AN "AS IS" BASIS, AND THE UNIVERSITY OF
CALIFORNIA HAS NO OBLIGATION TO PROVIDE MAINTENANCE, SUPPORT, UPDATES,
ENHANCEMENTS, OR MODIFICATIONS.
PT_COPYRIGHT_VERSION_2
COPYRIGHTENDKEY
*/
package ptolemy.media.javasound;
import java.io.IOException;
import java.net.URL;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
import javax.sound.sampled.UnsupportedAudioFileException;
//////////////////////////////////////////////////////////////////////////
//// SoundCapture
/**
A buffer supporting the capturing of audio samples from a file or
from the computer's audio input port.
<h2>Overview</h2>
A buffer supporting the capturing of audio samples from a file or
from the computer's audio input port. This class supports the
real-time capture of audio from the audio input port (mic or line-in)
as well as the capture of audio from a sound file specified as
a URL. Single channel
(mono) and multichannel audio (stereo) are supported. This class,
along with SoundPlayback, intends to provide an easy to use interface
to Java Sound, Java's audio API. Java Sound supports the capture
of audio data, but only at the byte level, which is audio format
specific. This class, however, provides higher level support for the
capture of double or integer valued samples from the computer's audio
input port or any supported sound file type. This class is therefore
useful when it one desires to capture audio samples in an audio format
independent way.
<p>
Depending on available audio
system resources, it may be possible to run an instance of this
class and an instance of SoundPlayback concurrently. This allows
for the concurrent capture, signal processing, and playback of audio data.
<p>
<h2>Usage</h2>
Two constructors are provided. One constructor creates a sound capture
object that captures from the line-in or microphone port.
The operating system must be used
to select between the microphone and line-in. This cannot be
done using Java. If this
constructor is used, there will be a small
delay between the time that the audio enters the microphone or
line-in and the time that the corresponding audio samples are
available via getSamples() or getSamplesInt().
This latency can be adjusted by setting the <i>bufferSize</i>
constructor parameter. Another constructor creates a sound capture
object that captures audio from a sound file specified as a URL.
<p>
After calling the appropriate constructor, startCapture()
must be called to initialize the audio system for capture.
The getSamples() or getSamplesInt() method should then be repeatedly
invoked to obtain audio data in the form of a multidimensional
array of audio sample values. getSamples() will return audio
sample values in the range [-1, 1]. getSamplesInt() will return
audio samples in the range
(-2^(bits_per_sample/2), 2^(bits_per_sample/2)), where
bits_per_sample is the number of bits per sample.
For the case where
audio is captured from the mic or line-in, it is important to
invoke getSamples() or getSamplesInt() often enough to prevent
overflow of
the internal audio buffer. The size of the internal buffer is
set in the constructor. Note that it is possible (but probably
not useful) to interleave calls to getSamples() and
getSamplesInt().
Finally, after no more audio data is desired, stopCapture()
should be called to free up audio system resources.
<p>
<h2>Security issues</h2>
Applications have no restrictions on the
capturing or playback of audio. Applets, however, may only capture
audio from a file specified as a URL on the same machine as the
one the applet was loaded from. Applet code is not allowed to
read or write native files. The .java.policy file must be
modified to grant applets more privileges.
<p>
Note: Requires Java 2 v1.3.0 or later.
@author Brian K. Vogel
@version $Id$
@since Ptolemy II 1.0
@Pt.ProposedRating Yellow (vogel)
@Pt.AcceptedRating Yellow (cxh)
@see ptolemy.media.javasound.SoundPlayback
*/
public class SoundCapture {
/** Construct a sound capture object that captures audio from a computer's
* audio input port. If this constructor is used, then it
* is important that getSamples() be
* invoked often enough to prevent overflow of the internal audio
* input buffer. Note the startCapture() must be called before the
* first invocation of getSamples(), otherwise getSamples() will
* throw an exception.
* @param sampleRate Sample rate in Hz. Must be in the range: 8000
* to 48000.
* @param sampleSizeInBits Number of bits per sample. Choices are
* 8 or 16.
* @param channels Number of audio channels. 1 for mono, 2 for
* stereo.
* @param bufferSize Requested size of the internal audio input
* buffer in samples. This controls the latency. A lower bound
* on the latency is given by (<i>bufferSize</i> / <i>sampleRate</i>)
* seconds. Ideally, the
* smallest value that gives acceptable performance (no overflow)
* should be used. Typical values are about 1/10 th the sample
* rate. For example, at 44100 Hz sample rate, a typical buffer
* size value might be 4410.
* @param getSamplesSize Size of the array returned by
* getSamples(). For performance reasons, the size should
* be chosen smaller than <i>bufferSize</i>. Typical values
* are 1/2 to 1/16th of <i>bufferSize</i>.
*/
public SoundCapture(float sampleRate, int sampleSizeInBits, int channels,
int bufferSize, int getSamplesSize) {
_isAudioCaptureActive = false;
// Set mode to real-time.
this._isRealTime = true;
this._sampleSizeInBits = sampleSizeInBits;
this._sampleRate = sampleRate;
this._channels = channels;
this._bufferSize = bufferSize;
this._productionRate = getSamplesSize;
}
/** Construct a sound capture object that captures audio from a
* sound file specified as a URL. Note that it is still possible
* to capture audio from a file on the local file system. For
* example, to capture from a sound file located at
* "C:\someDir\someFile.wave", <i>pathName</i>
* should be set to "file:///C:/someDir/someFile.wave".
* <p>
* Note the startCapture() must be called before the
* first invocation of getSamples(), otherwise getSamples() will
* throw an exception.
*
* @param pathName The name of the file as a URL. Valid sound file
* formats are WAVE (.wav), AIFF (.aif, .aiff), AU (.au). The file
* format is automatically determined from the file extension.
* If there is a problem reading the sound file, an IOException
* will be thrown in startCapture().
* @param getSamplesSize The number of samples per channel
* returned by getSamples().
*/
public SoundCapture(String pathName, int getSamplesSize) {
_isAudioCaptureActive = false;
// Set mode to "capture from file" (not real-time).
this._isRealTime = false;
this._pathName = pathName;
this._productionRate = getSamplesSize;
}
///////////////////////////////////////////////////////////////////
/// Public Methods ///
/** Return the number of audio channels. This method will
* return the number of audio channels, regardless of
* which constructor was used. However, this method is
* really only useful when the constructor that causes
* audio to be captured from a file is used, since
* the number of channels is unknown until the file
* is opened.
* <p>
* This method should
* be called while audio capture is active, i.e., after
* startCapture() is called and before stopCapture()
* is called.
*
* @return The number of audio channels. Return null if
* this method is called before startCapture().
*
* @exception IllegalStateException If this method is called
* before startCapture() is called or after stopCapture()
* is called.
*/
public int getChannels() throws IllegalStateException {
if (_isAudioCaptureActive == true) {
return _channels;
}
throw new IllegalStateException("SoundCapture: "
+ "getChannels() was called while audio capture was"
+ " inactive (startCapture() was never called).");
}
/** Return the sampling rate in Hz. This method will
* return the sampling rate, regardless of
* which constructor was used. However, this method is
* really only useful when the constructor that causes
* audio to be captured from a file is used, since
* the sampling rate is unknown until the file
* is opened.
* <p>
* This method should
* be called while audio capture is active, i.e., after
* startCapture() is called and before stopCapture()
* is called.
*
* @return The sample rate in Hz. Return null if
* this method is called before startCapture().
*
* @exception IllegalStateException If this method is called
* before startCapture() is called or after stopCapture()
* is called.
*/
public float getSampleRate() throws IllegalStateException {
if (_isAudioCaptureActive == true) {
return _sampleRate;
}
throw new IllegalStateException("SoundCapture: "
+ "getSampleRate() was called while audio capture was"
+ " inactive (startCapture() was never called).");
}
/** Return an array of captured audio samples. This method
* should be repeatedly called to obtain audio data.
* The returned audio samples will have values in the range
* [-1, 1], regardless of the audio bit resolution (bits per
* sample). When
* capturing from the computer's audio input port (mic or
* line-in), this method should be called often enough to
* prevent overflow of the internal audio buffer. If
* overflow occurs, some audio data will be lost but no
* exception or other error condition will occur. If
* the audio data is not yet available, then this method
* will block until the data is available. When capturing
* from a sound file, it is not possible for overflow to
* occur.
* <p>
* The array size
* is set by the <i>getSamplesSize</i> parameter in the
* constructor. For the case where audio is captured from
* the computer's audio-in port (mic or line-in), this
* method should be called often enough to prevent overflow
* of the internal audio buffer, the size of which is set
* in the constructor.
* @return Two dimensional array of captured audio samples.
* Return null
* if end of audio file is reached. A null return value is
* only possible when capturing from a sound file.
* The first index
* represents the channel number (0 for first channel, 1 for
* second channel, etc.). The second index represents the
* sample index within a channel. For example,
* <i>returned array</i>[n][m] contains the (m+1)th sample
* of the (n+1)th channel. For each channel, n, the length of
* <i>returned array</i>[n] is equal to <i>getSamplesSize</i>.
*
* @exception IOException If there is a problem capturing audio.
* @exception IllegalStateException If audio capture is currently
* inactive. That is, If startCapture() has not yet been called
* or if stopCapture() has already been called.
*/
public double[][] getSamples() throws IOException, IllegalStateException {
if (_isAudioCaptureActive == true) {
int numBytesRead;
if (_isRealTime == true) {
// Real-time capture.
numBytesRead = _targetLine.read(_data, 0, _productionRate
* _frameSizeInBytes);
} else {
// Capture audio from file.
numBytesRead = _properFormatAudioInputStream.read(_data);
}
if (numBytesRead == _data.length) {
// Convert byte array to double array.
_audioInDoubleArray = _byteArrayToDoubleArray(_data,
_bytesPerSample, _channels);
return _audioInDoubleArray;
} else if (numBytesRead != _data.length) {
// Read fewer samples than productionRate many samples.
// FIXME: There appears to be a java sound bug that
// causes AudioInputStream.read(array) to sometimes
// return fewer bytes than requested, even though
// the end of the file has not yet been reached.
_audioInDoubleArray = _byteArrayToDoubleArray(_data,
_bytesPerSample, _channels);
return _audioInDoubleArray;
} else if (numBytesRead == -1) {
// Ran out of samples to play. This generally means
// that the end of the sound file has been reached.
return null;
}
return null;
}
throw new IllegalStateException("SoundCapture: "
+ "getSamples() was called while audio capture was"
+ " inactive (startCapture() was never called or "
+ "stopCapture has already been called).");
}
/** Return an array of captured audio samples. This method
* should be repeatedly called to obtain audio data. This
* method requires less computation than getSamples(),
* since no conversion to doubles is performed. Therefore,
* the use of this method is recommended when integer
* valued audio samples are sufficient. The
* returned audio samples will have values in the range
* (-2^(bits_per_sample/2), 2^(bits_per_sample/2)). The
* range of sample values returned is therefore dependent
* on the bit resolution of the audio data. If this is not
* desired, then use getSamples() instead.
* <p>
* When capturing from the computer's audio input port (mic or
* line-in), this method should be called often enough to
* prevent overflow of the internal audio buffer. If
* overflow occurs, some audio data will be lost but no
* exception or other error condition will occur. If
* the audio data is not yet available, then this method
* will block until the data is available. When capturing
* from a sound file, it is not possible for overflow to
* occur.
* <p> The array size
* is set by the <i>getSamplesSize</i> parameter in the
* constructor. For the case where audio is captured from
* the computer's audio-in port (mic or line-in), this
* method should be called often enough to prevent overflow
* of the internal audio buffer, the size of which is set
* in the constructor.
* @return Two dimensional array of captured audio samples.
* Return null
* if end of audio file is reached A null return value is
* only possible when capturing from a sound file.
* The first index
* represents the channel number (0 for first channel, 1 for
* second channel, etc.). The second index represents the
* sample index within a channel. For example,
* <i>returned array</i>[n][m] contains the (m+1)th sample
* of the (n+1)th channel. For each channel, n, the length of
* <i>returned array</i>[n] is equal to <i>getSamplesSize</i>.
*
* @exception IOException If there is a problem capturing audio.
* @exception IllegalStateException If audio capture is currently
* inactive. That is, If startCapture() has not yet been called
* or if stopCapture() has already been called.
*/
public int[][] getSamplesInt() throws IOException, IllegalStateException {
if (_isAudioCaptureActive == true) {
int numBytesRead;
if (_isRealTime == true) {
// Real-time capture.
numBytesRead = _targetLine.read(_data, 0, _productionRate
* _frameSizeInBytes);
} else {
// Capture audio from file.
numBytesRead = _properFormatAudioInputStream.read(_data);
}
if (numBytesRead == _data.length) {
// Convert byte array to double array.
_audioInIntArray = _byteArrayToIntArray(_data, _bytesPerSample,
_channels);
return _audioInIntArray;
} else if (numBytesRead != _data.length) {
// Read fewer samples than productionRate many samples.
// FIXME: Output the samples that were read + zeros?
return null;
} else if (numBytesRead == -1) {
// Ran out of samples to play. This generally means
// that the end of the sound file has been reached.
return null;
}
return null;
}
throw new IllegalStateException("SoundCapture: "
+ "getSamples() was called while audio capture was"
+ " inactive (startCapture() was never called or "
+ "stopCapture has already been called).");
}
/** Begin capturing audio. This method must be invoked prior
* to the first invocation of getSamples(). If this is not
* done, then getSamples() will throw an exception when
* it is invoked. It is safe
* to call getSamples() immediately after this method returns.
* This method must not be called more than
* once between invocations of stopCapture(). Calling
* this method more than once between invocations of
* stopCapture() will cause this method to throw an exception.
*
* @exception IOException If there is a problem setting up
* the system for audio capture. This will occur if the
* a URL cannot be opened or if the audio in port cannot
* be accessed.
* @exception IllegalStateException If this method is called
* more than once between invocations of stopCapture().
*/
public void startCapture() throws IOException, IllegalStateException {
if (_isAudioCaptureActive == false) {
// FIXME: check and throw Exceptions
if (_isRealTime == true) {
_startCaptureRealTime();
} else {
_startCaptureFromFile();
}
_isAudioCaptureActive = true;
} else {
throw new IllegalStateException("SoundCapture: "
+ "startCapture() was called while audio capture was"
+ " already active (startCapture() was called "
+ "more than once between invocations of stopCapture()).");
}
}
/** Stop capturing audio. This method should be called when
* no more calls to getSamples(). are required, so
* that the system resources involved in the audio capture
* may be freed.
*
* @exception IOException If there is a problem closing the
* audio resources.
*/
public void stopCapture() throws IOException {
if (_isAudioCaptureActive == true) {
// Free up audio system resources.
// For capture from file:
if (_audioInputStream != null) {
_audioInputStream.close();
// FIXME : is this correct?
_audioInputStream = null;
}
if (_properFormatAudioInputStream != null) {
_properFormatAudioInputStream.close();
// FIXME : is this correct?
_properFormatAudioInputStream = null;
}
// For real-time capture:
if (_targetLine != null) {
if (_targetLine.isOpen() == true) {
_targetLine.stop();
_targetLine.close();
_targetLine = null;
}
}
}
_isAudioCaptureActive = false;
}
/** Return the number of bits per audio sample. This method will
* return the number of bits per audio sample, regardless of
* which constructor was used. However, this method is
* really only useful when the constructor that causes
* audio to be captured from a file is used, since
* the number of bits per audio sample is unknown until the file
* is opened.
* <p>
* This method must
* be called while audio capture is active, i.e., after
* startCapture() is called and before stopCapture()
* is called, or else an exception will be thrown.
*
* @return The sample size in bits. Return null if
* this method is called before startCapture().
*
* @exception IllegalStateException If this method is called
* before startCapture() is called or after stopCapture()
* is called.
*/
public int getSampleSizeInBits() throws IllegalStateException {
if (_isAudioCaptureActive == true) {
return _sampleSizeInBits;
}
throw new IllegalStateException("SoundCapture: "
+ "getSampleSizeInBits() was called while audio capture was"
+ " inactive (startCapture() was never called).");
}
///////////////////////////////////////////////////////////////////
//// private methods ////
private void _startCaptureRealTime() throws IOException {
boolean signed = true;
boolean bigEndian = true;
AudioFormat format = new AudioFormat(_sampleRate, _sampleSizeInBits,
_channels, signed, bigEndian);
_frameSizeInBytes = format.getFrameSize();
DataLine.Info targetInfo = new DataLine.Info(TargetDataLine.class,
format, AudioSystem.NOT_SPECIFIED);
// The following works under Windows Java 1.3.0 RC2 but
// not under Tritonus under Linux, so comment out.
//if (!AudioSystem.isLineSupported(targetInfo)) {
// // FIXME: throw exception here.
// System.out.println("Line matching " + targetInfo +
// " not supported.");
// return;
//}
try {
_targetLine = (TargetDataLine) AudioSystem.getLine(targetInfo);
// Note: 2nd parameter is the buffer size (in bytes).
// Larger values increase latency but may be required if
// garbage collection, etc. is an issue.
_targetLine.open(format, _bufferSize * _frameSizeInBytes);
} catch (LineUnavailableException ex) {
throw new IOException("Unable to open the line for "
+ "real-time audio capture: " + ex);
}
// The following works under Windows Java 1.3.0 RC2 but
// not under Tritonus under Linux, so comment out.
//if (!AudioSystem.isLineSupported(sourceInfo)) {
// //FIXME: handle this correctly.
// System.err.println("Line matching " + sourceInfo +
// " not supported.");
// return;
//}
// Array of audio samples in byte format.
_data = new byte[_productionRate * _frameSizeInBytes];
_bytesPerSample = _sampleSizeInBits / 8;
// Start the target data line
_targetLine.start();
}
/* Perform necessary initialization to capture from a sound
* file. The sound file is specified as a URL.
*/
private void _startCaptureFromFile() throws IOException {
// Load audio from a URL.
// Create a URL corresponding to the sound file location.
URL soundURL = new URL(_pathName);
if (soundURL != null) {
try {
_audioInputStream = AudioSystem.getAudioInputStream(soundURL);
} catch (UnsupportedAudioFileException e) {
throw new IOException("Unsupported AudioFile :" + e);
}
}
// make sure we have something to play
if (_audioInputStream == null) {
throw new IOException("No loaded audio to play back");
}
// FIXME: is this correct?
//_audioInputStream.reset();
AudioFormat origFormat = _audioInputStream.getFormat();
// Now convert to PCM_SIGNED_BIG_ENDIAN so that can get double
// representation of samples.
float sampleRate = origFormat.getSampleRate();
_sampleSizeInBits = origFormat.getSampleSizeInBits();
_bytesPerSample = _sampleSizeInBits / 8;
_channels = origFormat.getChannels();
boolean signed = true;
boolean bigEndian = true;
AudioFormat format = new AudioFormat(sampleRate, _sampleSizeInBits,
_channels, signed, bigEndian);
_properFormatAudioInputStream = AudioSystem.getAudioInputStream(format,
_audioInputStream);
_frameSizeInBytes = format.getFrameSize();
// FIXME: is this correct?
//_properFormatAudioInputStream.reset();
// Array of audio samples in byte format.
_data = new byte[_productionRate * _frameSizeInBytes];
}
/* Convert a byte array of audio samples in linear signed pcm big endian
* format into a double array of audio samples (-1, 1) range.
* @param byteArray The linear signed pcm big endian byte array
* formatted array representation of audio data.
* @param bytesPerSample Number of bytes per sample. Supported
* bytes per sample by this method are 8, 16, 24, 32.
* @param channels Number of audio channels. 1 for mono, 2 for
* stereo.
* @return Two dimensional array holding audio samples.
* For each channel, m, doubleArray[m] is a single dimensional
* array containing samples for channel m.
*/
private double[][] _byteArrayToDoubleArray(byte[] byteArray,
int bytesPerSample, int channels) {
int lengthInSamples = byteArray.length / (bytesPerSample * channels);
// Check if we need to reallocate.
if ((channels != _doubleArray.length)
|| (lengthInSamples != _doubleArray[0].length)) {
// Reallocate
_doubleArray = new double[channels][lengthInSamples];
}
//double maxSampleReciprocal = 1/(Math.pow(2, 8 * bytesPerSample - 1));
// Could use above line, but hopefully, code below will
// be faster.
double maxSampleReciprocal;
if (bytesPerSample == 2) {
// 1 / 32768
maxSampleReciprocal = 3.0517578125e-5;
} else if (bytesPerSample == 1) { // 1 / 128
maxSampleReciprocal = 7.8125e-3;
} else if (bytesPerSample == 3) {
// 1 / 8388608
maxSampleReciprocal = 1.1920928955e07;
} else if (bytesPerSample == 4) {
// 1 / 147483648e9
maxSampleReciprocal = 4.655661287308e-10;
} else {
// Should not happen.
maxSampleReciprocal = 0;
}
// Check if we need to reallocate.
// FIXME: This test is really not needed since bytesPerSample
// is set in the constructor. It should never change.
if (bytesPerSample != _b.length) {
_b = new byte[bytesPerSample];
}
for (int currSamp = 0; currSamp < lengthInSamples; currSamp++) {
// For each channel,
for (int currChannel = 0; currChannel < channels; currChannel++) {
for (int i = 0; i < bytesPerSample; i += 1) {
// Assume we are dealing with big endian.
_b[i] = byteArray[(currSamp * bytesPerSample * channels)
+ (bytesPerSample * currChannel) + i];
}
int result = (_b[0] >> 7);
for (int i = 0; i < bytesPerSample; i += 1) {
result = (result << 8) + (_b[i] & 0xff);
}
_doubleArray[currChannel][currSamp] = result
* maxSampleReciprocal;
}
}
return _doubleArray;
}
/* Convert a byte array of audio samples in linear signed pcm big endian
* format into a (signed) int array of audio samples. The range
* of the returned samples is approximately
* (-2^(bits_per_sample/2), 2^(bits_per_sample/2)).
* @param byteArray The linear signed pcm big endian byte array
* formatted array representation of audio data.
* @param bytesPerSample Number of bytes per sample. Supported
* bytes per sample by this method are 8, 16, 24, 32.
* @param channels Number of audio channels. 1 for mono, 2 for
* stereo.
* @return Two dimensional array holding audio samples.
* For each channel, m, intArray[m] is a single dimensional
* array containing samples for channel m.
*/
private int[][] _byteArrayToIntArray(byte[] byteArray, int bytesPerSample,
int channels) {
int lengthInSamples = byteArray.length / (bytesPerSample * channels);
// Check if we need to reallocate.
if ((channels != _doubleArray.length)
|| (lengthInSamples != _doubleArray[0].length)) {
// Reallocate
_intArray = new int[channels][lengthInSamples];
}
// Check if we need to reallocate.
// FIXME: This test is really not needed since bytesPerSample
// is set in the constructor. It should never change.
if (bytesPerSample != _b.length) {
_b = new byte[bytesPerSample];
}
for (int currSamp = 0; currSamp < lengthInSamples; currSamp++) {
// For each channel,
for (int currChannel = 0; currChannel < channels; currChannel++) {
for (int i = 0; i < bytesPerSample; i += 1) {
// Assume we are dealing with big endian.
_b[i] = byteArray[(currSamp * bytesPerSample * channels)
+ (bytesPerSample * currChannel) + i];
}
int result = (_b[0] >> 7);
for (int i = 0; i < bytesPerSample; i += 1) {
result = (result << 8) + (_b[i] & 0xff);
}
_intArray[currChannel][currSamp] = result;
}
}
return _intArray;
}
///////////////////////////////////////////////////////////////////
//// private variables ////
private AudioInputStream _properFormatAudioInputStream;
private AudioInputStream _audioInputStream;
private int _productionRate;
// Array of audio samples in double format.
private double[][] _audioInDoubleArray;
// Array of audio samples in int format.
private int[][] _audioInIntArray;
// Array of audio samples in byte format.
private byte[] _data;
private int _frameSizeInBytes;
private boolean _isRealTime;
private String _pathName;
private int _sampleSizeInBits;
private float _sampleRate;
private int _channels;
private int _bufferSize;
private TargetDataLine _targetLine;
private int _bytesPerSample;
private boolean _isAudioCaptureActive;
private byte[] _b = new byte[1];
private double[][] _doubleArray = new double[1][1];
private int[][] _intArray = new int[1][1];
}