我是Java的新手，虽然我已经掌握了语法和结构，但我很难从麦克风中获取处理过的数字音频样本。
我想要实现的是非常简单的，而从长远来看，我正在尝试创建一个非常简单的频谱图，但只是为了理解/掌握我试图从头开始的音频处理过程。
这是我的问题
当麦克风检测到一声嘟嘟声或任何声音时，我想捕获那些二进制数据，并只是以原始格式显示它。
来自JAVA真的太过分了吗？
我已经阅读了有关模拟/数字信号，FFT，matlab的内容，我搜索了许多链接，如下所示：
Is there a way of recording audio of streaming broadcast from a webpage?
OpenAL playback captured audio data c++
和oracle着名的介绍
http://docs.oracle.com/javase/tutorial/sound/capturing.html
但他们都无法为我的答案提供解决方案。
我不是要求代码，虽然它会很棒，只是阅读每一行并理解所涉及的机制，但一个简单的提示也会很好。
这是一个简单的代码，用于捕获字节，但仅限于现有的wav文件
import java.io.FileInputStream;
import java.io.IOException;
public class Boo{
public static void main(String[] arguments){
try {
FileInputStream file = new FileInputStream("beep.wav");
boolean eof = false;
int count = 0;
while(!eof){
int input = file.read();
System.out.print(input + "");
if(input == -1)
eof = true;
else
count++;
}
file.close();
System.out.println("\nBytes read: " + count);
}catch (IOException e){
System.out.println("Error - " + e.toString());
}
}
}
- 为了更清晰 -
所有我试图让它只是一个简单的程序来读取麦克风。并显示它实时捕获的声音的二进制数据。
将其想象为频谱图，当捕获声音时，图形会根据信号电平的变化而上下移动，但在这种情况下，不需要将二进制数据转换为音频图形，只需要显示任何原始数据本身。无需写/读文件。只需从麦克风中捕捉，并显示从麦克风中读取的内容。
如果上面提到的很难，因为我在网上搜索过，找不到任何有用的东西，你可以给我指导/指示.. 感谢
答案 0 :(得分：8)
javax.sound.sampled包应该包含您所需要的一切。
示例：
int duration = 5; // sample for 5 seconds
TargetDataLine line = null;
// find a DataLine that can be read
// (maybe hardcode this if you have multiple microphones)
Info[] mixerInfo = AudioSystem.getMixerInfo();
for (int i = 0; i < mixerInfo.length; i++) {
Mixer mixer = AudioSystem.getMixer(mixerInfo[i]);
Line.Info[] targetLineInfo = mixer.getTargetLineInfo();
if (targetLineInfo.length > 0) {
line = (TargetDataLine) mixer.getLine(targetLineInfo[0]);
break;
}
}
if (line == null)
throw new UnsupportedOperationException("No recording device found");
AudioFormat af = new AudioFormat(11000, 8, 1, true, false);
line.open(af);
line.start();
ByteArrayOutputStream baos = new ByteArrayOutputStream();
byte[] buf = new byte[(int)af.getSampleRate() * af.getFrameSize()];
long end = System.currentTimeMillis() + 1000 * duration;
int len;
while (System.currentTimeMillis() < end && ((len = line.read(buf, 0, buf.length)) != -1)) {
baos.write(buf, 0, len);
}
line.stop();
line.close();
baos.close();
之后，您可以从字节数组输出流中挖出字节。或者，如果您愿意，您当然可以在while循环内处理它们。
答案 1 :(得分：3)
以下是从Sphinx4麦克风支持中复制粘贴的一些代码。我希望它会有用。
指向Sphinx主页的链接：http://cmusphinx.sourceforge.net/sphinx4/。
/**
* <p/> A Microphone captures audio data from the system's underlying audio input systems. Converts these audio data
* into Data objects. When the method <code>startRecording()</code> is called, a new thread will be created and used to
* capture audio, and will stop when <code>stopRecording()</code> is called. Calling <code>getData()</code> returns the
* captured audio data as Data objects. </p> <p/> This Microphone will attempt to obtain an audio device with the format
* specified in the configuration. If such a device with that format cannot be obtained, it will try to obtain a device
* with an audio format that has a higher sample rate than the configured sample rate, while the other parameters of the
* format (i.e., sample size, endianness, sign, and channel) remain the same. If, again, no such device can be obtained,
* it flags an error, and a call <code>startRecording</code> returns false. </p>
*/
public class Microphone extends BaseDataProcessor {
/**
* The property for the sample rate of the data.
*/
@S4Integer(defaultValue = 16000)
public static final String PROP_SAMPLE_RATE = "sampleRate";
/**
* The property that specifies whether or not the microphone will release the audio between utterances. On
* certain systems (Linux for one), closing and reopening the audio does not work too well. The default is false for
* Linux systems, true for others.
*/
@S4Boolean(defaultValue = true)
public final static String PROP_CLOSE_BETWEEN_UTTERANCES = "closeBetweenUtterances";
/**
* The property that specifies the number of milliseconds of audio data to read each time from the underlying
* Java Sound audio device.
*/
@S4Integer(defaultValue = 10)
public final static String PROP_MSEC_PER_READ = "msecPerRead";
/**
* The property for the number of bits per value.
*/
@S4Integer(defaultValue = 16)
public static final String PROP_BITS_PER_SAMPLE = "bitsPerSample";
/**
* The property specifying the number of channels.
*/
@S4Integer(defaultValue = 1)
public static final String PROP_CHANNELS = "channels";
/**
* The property specify the endianness of the data.
*/
@S4Boolean(defaultValue = true)
public static final String PROP_BIG_ENDIAN = "bigEndian";
/**
* The property specify whether the data is signed.
*/
@S4Boolean(defaultValue = true)
public static final String PROP_SIGNED = "signed";
/**
* The property that specifies whether to keep the audio data of an utterance around until the next utterance
* is recorded.
*/
@S4Boolean(defaultValue = false)
public final static String PROP_KEEP_LAST_AUDIO = "keepLastAudio";
/**
* The property that specifies how to convert stereo audio to mono. Currently, the possible values are
* "average", which averages the samples from at each channel, or "selectChannel", which chooses audio only from
* that channel. If you choose "selectChannel", you should also specify which channel to use with the
* "selectChannel" property.
*/
@S4String(defaultValue = "average", range = {"average", "selectChannel"})
public final static String PROP_STEREO_TO_MONO = "stereoToMono";
/**
* The property that specifies the channel to use if the audio is stereo
*/
@S4Integer(defaultValue = 0)
public final static String PROP_SELECT_CHANNEL = "selectChannel";
/**
* The property that specifies the mixer to use. The value can be "default," (which means let the
* AudioSystem decide), "last," (which means select the last Mixer supported by the AudioSystem), which appears to
* be what is often used for USB headsets, or an integer value which represents the index of the Mixer.Info that is
* returned by AudioSystem.getMixerInfo(). To get the list of Mixer.Info objects, run the AudioTool application with
* a command line argument of "-dumpMixers".
*
* @see edu.cmu.sphinx.tools.audio.AudioTool
*/
@S4String(defaultValue = "default")
public final static String PROP_SELECT_MIXER = "selectMixer";
private AudioFormat finalFormat;
private AudioInputStream audioStream;
private TargetDataLine audioLine;
private BlockingQueue<Data> audioList;
private Utterance currentUtterance;
private boolean doConversion;
private final int audioBufferSize = 160000;
private volatile boolean recording;
private volatile boolean utteranceEndReached = true;
private RecordingThread recorder;
// Configuration data
private AudioFormat desiredFormat;
private Logger logger;
private boolean closeBetweenUtterances;
private boolean keepDataReference;
private boolean signed;
private boolean bigEndian;
private int frameSizeInBytes;
private int msecPerRead;
private int selectedChannel;
private String selectedMixerIndex;
private String stereoToMono;
private int sampleRate;
/**
* @param sampleRate sample rate of the data
* @param bitsPerSample number of bits per value.
* @param channels number of channels.
* @param bigEndian the endianness of the data
* @param signed whether the data is signed.
* @param closeBetweenUtterances whether or not the microphone will release the audio between utterances. On
* certain systems (Linux for one), closing and reopening the audio does not work too well. The default is false for
* Linux systems, true for others
* @param msecPerRead the number of milliseconds of audio data to read each time from the underlying
* Java Sound audio device.
* @param keepLastAudio whether to keep the audio data of an utterance around until the next utterance
* is recorded.
* @param stereoToMono how to convert stereo audio to mono. Currently, the possible values are
* "average", which averages the samples from at each channel, or "selectChannel", which chooses audio only from
* that channel. If you choose "selectChannel", you should also specify which channel to use with the
* "selectChannel" property.
* @param selectedChannel the channel to use if the audio is stereo
* @param selectedMixerIndex the mixer to use. The value can be "default," (which means let the
* AudioSystem decide), "last," (which means select the last Mixer supported by the AudioSystem), which appears to
* be what is often used for USB headsets, or an integer value which represents the index of the Mixer.Info that is
* returned by AudioSystem.getMixerInfo(). To get the list of Mixer.Info objects, run the AudioTool application with
* a command line argument of "-dumpMixers".
*/
public Microphone(int sampleRate, int bitsPerSample, int channels,
boolean bigEndian, boolean signed, boolean closeBetweenUtterances, int msecPerRead, boolean keepLastAudio,
String stereoToMono, int selectedChannel, String selectedMixerIndex) {
initLogger();
this.bigEndian = bigEndian;
this.signed = signed;
this.desiredFormat = new AudioFormat
((float) sampleRate, bitsPerSample, channels, signed, bigEndian);
this.closeBetweenUtterances = closeBetweenUtterances;
this.msecPerRead = msecPerRead;
this.keepDataReference = keepLastAudio;
this.stereoToMono = stereoToMono;
this.selectedChannel = selectedChannel;
this.selectedMixerIndex = selectedMixerIndex;
}
public Microphone() {
}
/*
* (non-Javadoc)
*
* @see edu.cmu.sphinx.util.props.Configurable#newProperties(edu.cmu.sphinx.util.props.PropertySheet)
*/
@Override
public void newProperties(PropertySheet ps) throws PropertyException {
super.newProperties(ps);
logger = ps.getLogger();
sampleRate = ps.getInt(PROP_SAMPLE_RATE);
int sampleSizeInBits = ps.getInt(PROP_BITS_PER_SAMPLE);
int channels = ps.getInt(PROP_CHANNELS);
bigEndian = ps.getBoolean(PROP_BIG_ENDIAN);
signed = ps.getBoolean(PROP_SIGNED);
desiredFormat = new AudioFormat
((float) sampleRate, sampleSizeInBits, channels, signed, bigEndian);
closeBetweenUtterances = ps.getBoolean(PROP_CLOSE_BETWEEN_UTTERANCES);
msecPerRead = ps.getInt(PROP_MSEC_PER_READ);
keepDataReference = ps.getBoolean(PROP_KEEP_LAST_AUDIO);
stereoToMono = ps.getString(PROP_STEREO_TO_MONO);
selectedChannel = ps.getInt(PROP_SELECT_CHANNEL);
selectedMixerIndex = ps.getString(PROP_SELECT_MIXER);
}
/**
* Constructs a Microphone with the given InputStream.
*/
@Override
public void initialize() {
super.initialize();
audioList = new LinkedBlockingQueue<Data>();
DataLine.Info info
= new DataLine.Info(TargetDataLine.class, desiredFormat);
/* If we cannot get an audio line that matches the desired
* characteristics, shoot for one that matches almost
* everything we want, but has a higher sample rate.
*/
if (!AudioSystem.isLineSupported(info)) {
logger.info(desiredFormat + " not supported");
AudioFormat nativeFormat
= DataUtil.getNativeAudioFormat(desiredFormat,
getSelectedMixer());
if (nativeFormat == null) {
logger.severe("couldn't find suitable target audio format");
} else {
finalFormat = nativeFormat;
/* convert from native to the desired format if supported */
doConversion = AudioSystem.isConversionSupported
(desiredFormat, nativeFormat);
if (doConversion) {
logger.info
("Converting from " + finalFormat.getSampleRate()
+ "Hz to " + desiredFormat.getSampleRate() + "Hz");
} else {
logger.info
("Using native format: Cannot convert from " +
finalFormat.getSampleRate() + "Hz to " +
desiredFormat.getSampleRate() + "Hz");
}
}
} else {
logger.info("Desired format: " + desiredFormat + " supported.");
finalFormat = desiredFormat;
}
}
/**
* Gets the Mixer to use. Depends upon selectedMixerIndex being defined.
*
* @see #newProperties
*/
private Mixer getSelectedMixer() {
if (selectedMixerIndex.equals("default")) {
return null;
} else {
Mixer.Info[] mixerInfo = AudioSystem.getMixerInfo();
if (selectedMixerIndex.equals("last")) {
return AudioSystem.getMixer(mixerInfo[mixerInfo.length - 1]);
} else {
int index = Integer.parseInt(selectedMixerIndex);
return AudioSystem.getMixer(mixerInfo[index]);
}
}
}
/**
* Creates the audioLine if necessary and returns it.
*/
private TargetDataLine getAudioLine() {
if (audioLine != null) {
return audioLine;
}
/* Obtain and open the line and stream.
*/
try {
/* The finalFormat was decided in the initialize() method
* and is based upon the capabilities of the underlying
* audio system. The final format will have all the
* desired audio characteristics, but may have a sample
* rate that is higher than desired. The idea here is
* that we'll let the processors in the front end (e.g.,
* the FFT) handle some form of downsampling for us.
*/
logger.info("Final format: " + finalFormat);
DataLine.Info info = new DataLine.Info(TargetDataLine.class,
finalFormat);
/* We either get the audio from the AudioSystem (our
* default choice), or use a specific Mixer if the
* selectedMixerIndex property has been set.
*/
Mixer selectedMixer = getSelectedMixer();
if (selectedMixer == null) {
audioLine = (TargetDataLine) AudioSystem.getLine(info);
} else {
audioLine = (TargetDataLine) selectedMixer.getLine(info);
}
/* Add a line listener that just traces
* the line states.
*/
audioLine.addLineListener(new LineListener() {
@Override
public void update(LineEvent event) {
logger.info("line listener " + event);
}
});
} catch (LineUnavailableException e) {
logger.severe("microphone unavailable " + e.getMessage());
}
return audioLine;
}
/**
* Opens the audio capturing device so that it will be ready for capturing audio. Attempts to create a converter if
* the requested audio format is not directly available.
*
* @return true if the audio capturing device is opened successfully; false otherwise
*/
private boolean open() {
TargetDataLine audioLine = getAudioLine();
if (audioLine != null) {
if (!audioLine.isOpen()) {
logger.info("open");
try {
audioLine.open(finalFormat, audioBufferSize);
} catch (LineUnavailableException e) {
logger.severe("Can't open microphone " + e.getMessage());
return false;
}
audioStream = new AudioInputStream(audioLine);
if (doConversion) {
audioStream = AudioSystem.getAudioInputStream
(desiredFormat, audioStream);
assert (audioStream != null);
}
/* Set the frame size depending on the sample rate.
*/
float sec = ((float) msecPerRead) / 1000.f;
frameSizeInBytes =
(audioStream.getFormat().getSampleSizeInBits() / 8) *
(int) (sec * audioStream.getFormat().getSampleRate());
logger.info("Frame size: " + frameSizeInBytes + " bytes");
}
return true;
} else {
logger.severe("Can't find microphone");
return false;
}
}
/**
* Returns the format of the audio recorded by this Microphone. Note that this might be different from the
* configured format.
*
* @return the current AudioFormat
*/
public AudioFormat getAudioFormat() {
return finalFormat;
}
/**
* Returns the current Utterance.
*
* @return the current Utterance
*/
public Utterance getUtterance() {
return currentUtterance;
}
/**
* Returns true if this Microphone is recording.
*
* @return true if this Microphone is recording, false otherwise
*/
public boolean isRecording() {
return recording;
}
/**
* Starts recording audio. This method will return only when a START event is received, meaning that this Microphone
* has started capturing audio.
*
* @return true if the recording started successfully; false otherwise
*/
public synchronized boolean startRecording() {
if (recording) {
return false;
}
if (!open()) {
return false;
}
utteranceEndReached = false;
if (audioLine.isRunning()) {
logger.severe("Whoops: audio line is running");
}
assert (recorder == null);
recorder = new RecordingThread("Microphone");
recorder.start();
recording = true;
return true;
}
/**
* Stops recording audio. This method does not return until recording has been stopped and all data has been read
* from the audio line.
*/
public synchronized void stopRecording() {
if (audioLine != null) {
if (recorder != null) {
recorder.stopRecording();
recorder = null;
}
recording = false;
}
}
/**
* This Thread records audio, and caches them in an audio buffer.
*/
class RecordingThread extends Thread {
private boolean done;
private volatile boolean started;
private long totalSamplesRead;
private final Object lock = new Object();
/**
* Creates the thread with the given name
*
* @param name the name of the thread
*/
public RecordingThread(String name) {
super(name);
}
/**
* Starts the thread, and waits for recorder to be ready
*/
@Override
public void start() {
started = false;
super.start();
waitForStart();
}
/**
* Stops the thread. This method does not return until recording has actually stopped, and all the data has been
* read from the audio line.
*/
public void stopRecording() {
audioLine.stop();
try {
synchronized (lock) {
while (!done) {
lock.wait();
}
}
} catch (InterruptedException e) {
e.printStackTrace();
}
// flush can not be called here because the audio-line might has been set to null already by the mic-thread
// audioLine.flush();
}
/**
* Implements the run() method of the Thread class. Records audio, and cache them in the audio buffer.
*/
@Override
public void run() {
totalSamplesRead = 0;
logger.info("started recording");
if (keepDataReference) {
currentUtterance = new Utterance
("Microphone", audioStream.getFormat());
}
audioList.add(new DataStartSignal(sampleRate));
logger.info("DataStartSignal added");
try {
audioLine.start();
while (!done) {
Data data = readData(currentUtterance);
if (data == null) {
done = true;
break;
}
audioList.add(data);
}
audioLine.flush();
if (closeBetweenUtterances) {
/* Closing the audio stream *should* (we think)
* also close the audio line, but it doesn't
* appear to do this on the Mac. In addition,
* once the audio line is closed, re-opening it
* on the Mac causes some issues. The Java sound
* spec is also kind of ambiguous about whether a
* closed line can be re-opened. So...we'll go
* for the conservative route and never attempt
* to re-open a closed line.
*/
audioStream.close();
audioLine.close();
System.err.println("set to null");
audioLine = null;
}
} catch (IOException ioe) {
logger.warning("IO Exception " + ioe.getMessage());
ioe.printStackTrace();
}
long duration = (long)
(((double) totalSamplesRead /
(double) audioStream.getFormat().getSampleRate()) * 1000.0);
audioList.add(new DataEndSignal(duration));
logger.info("DataEndSignal ended");
logger.info("stopped recording");
synchronized (lock) {
lock.notify();
}
}
/**
* Waits for the recorder to start
*/
private synchronized void waitForStart() {
// note that in theory we coulde use a LineEvent START
// to tell us when the microphone is ready, but we have
// found that some javasound implementations do not always
// issue this event when a line is opened, so this is a
// WORKAROUND.
try {
while (!started) {
wait();
}
} catch (InterruptedException ie) {
logger.warning("wait was interrupted");
}
}
/**
* Reads one frame of audio data, and adds it to the given Utterance.
*
* @param utterance
* @return an Data object containing the audio data
* @throws java.io.IOException
*/
private Data readData(Utterance utterance) throws IOException {
// Read the next chunk of data from the TargetDataLine.
byte[] data = new byte[frameSizeInBytes];
int channels = audioStream.getFormat().getChannels();
long collectTime = System.currentTimeMillis();
long firstSampleNumber = totalSamplesRead / channels;
int numBytesRead = audioStream.read(data, 0, data.length);
// notify the waiters upon start
if (!started) {
synchronized (this) {
started = true;
notifyAll();
}
}
if (logger.isLoggable(Level.FINE)) {
logger.info("Read " + numBytesRead
+ " bytes from audio stream.");
}
if (numBytesRead <= 0) {
return null;
}
int sampleSizeInBytes =
audioStream.getFormat().getSampleSizeInBits() / 8;
totalSamplesRead += (numBytesRead / sampleSizeInBytes);
if (numBytesRead != frameSizeInBytes) {
if (numBytesRead % sampleSizeInBytes != 0) {
throw new Error("Incomplete sample read.");
}
data = Arrays.copyOf(data, numBytesRead);
}
if (keepDataReference) {
utterance.add(data);
}
double[] samples;
if (bigEndian) {
samples = DataUtil.bytesToValues
(data, 0, data.length, sampleSizeInBytes, signed);
} else {
samples = DataUtil.littleEndianBytesToValues
(data, 0, data.length, sampleSizeInBytes, signed);
}
if (channels > 1) {
samples = convertStereoToMono(samples, channels);
}
return (new DoubleData
(samples, (int) audioStream.getFormat().getSampleRate(),
collectTime, firstSampleNumber));
}
}
/**
* Converts stereo audio to mono.
*
* @param samples the audio samples, each double in the array is one sample
* @param channels the number of channels in the stereo audio
*/
private double[] convertStereoToMono(double[] samples, int channels) {
assert (samples.length % channels == 0);
double[] finalSamples = new double[samples.length / channels];
if (stereoToMono.equals("average")) {
for (int i = 0, j = 0; i < samples.length; j++) {
double sum = samples[i++];
for (int c = 1; c < channels; c++) {
sum += samples[i++];
}
finalSamples[j] = sum / channels;
}
} else if (stereoToMono.equals("selectChannel")) {
for (int i = selectedChannel, j = 0; i < samples.length;
i += channels, j++) {
finalSamples[j] = samples[i];
}
} else {
throw new Error("Unsupported stereo to mono conversion: " +
stereoToMono);
}
return finalSamples;
}
/**
* Clears all cached audio data.
*/
public void clear() {
audioList.clear();
}
/**
* Reads and returns the next Data object from this Microphone, return null if there is no more audio data. All
* audio data captured in-between <code>startRecording()</code> and <code>stopRecording()</code> is cached in an
* Utterance object. Calling this method basically returns the next chunk of audio data cached in this Utterance.
*
* @return the next Data or <code>null</code> if none is available
*/
@Override
public Data getData() throws DataProcessingException {
getTimer().start();
Data output = null;
if (!utteranceEndReached) {
try {
output = audioList.take();
} catch (InterruptedException ie) {
throw new DataProcessingException("cannot take Data from audioList", ie);
}
if (output instanceof DataEndSignal) {
utteranceEndReached = true;
}
}
getTimer().stop();
// signalCheck(output);
return output;
}
/**
* Returns true if there is more data in the Microphone.
* This happens either if the a DataEndSignal data was not taken from the buffer,
* or if the buffer in the Microphone is not yet empty.
*
* @return true if there is more data in the Microphone
*/
public boolean hasMoreData() {
return !(utteranceEndReached && audioList.isEmpty());
}
}
答案 2 :(得分：2)
看看xuggle。他们放弃了对该项目的支持，但它仍然拥有所有文档，而google-group有时会给你一个很好的答案。
至于专门从硬件中读取音频，请从这些demos开始，然后从那里开始工作。
最困难的部分是从麦克风中流式传输数据，这在您链接的article中得到了很好的解释。
然后使用oracle文档将其流式传输到xuggle中，然后按照您认为合适的方式操作音频。
答案 3 :(得分：2)