我正在实现从MIC到另一台PC的java服务器的实时流媒体。但我只听到白噪声。
我已经附加了客户端和服务器程序
Client:
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.SocketException;
import java.net.UnknownHostException;
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;
public class Mic
{
public byte[] buffer;
private int port;
static AudioInputStream ais;
public static void main(String[] args)
{
TargetDataLine line;
DatagramPacket dgp;
AudioFormat.Encoding encoding = AudioFormat.Encoding.PCM_SIGNED;
float rate = 44100.0f;
int channels = 2;
int sampleSize = 16;
boolean bigEndian = true;
InetAddress addr;
AudioFormat format = new AudioFormat(encoding, rate, sampleSize, channels, (sampleSize / 8) * channels, rate, bigEndian);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
if (!AudioSystem.isLineSupported(info)) {
System.out.println("Line matching " + info + " not supported.");
return;
}
try
{
line = (TargetDataLine) AudioSystem.getLine(info);
int buffsize = line.getBufferSize()/5;
buffsize += 512;
line.open(format);
line.start();
int numBytesRead;
byte[] data = new byte[buffsize];
addr = InetAddress.getByName("127.0.0.1");
DatagramSocket socket = new DatagramSocket();
while (true) {
// Read the next chunk of data from the TargetDataLine.
numBytesRead = line.read(data, 0, data.length);
// Save this chunk of data.
dgp = new DatagramPacket (data,data.length,addr,50005);
socket.send(dgp);
}
}catch (LineUnavailableException e) {
e.printStackTrace();
}catch (UnknownHostException e) {
// TODO: handle exception
} catch (SocketException e) {
// TODO: handle exception
} catch (IOException e2) {
// TODO: handle exception
}
}
}
并且服务器端没有问题。它与Android客户端AudioRecord完美运行。
Server:
import java.io.ByteArrayInputStream;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.SourceDataLine;
public class Server {
AudioInputStream audioInputStream;
static AudioInputStream ais;
static AudioFormat format;
static boolean status = true;
static int port = 50005;
static int sampleRate = 44100;
static DataLine.Info dataLineInfo;
static SourceDataLine sourceDataLine;
public static void main(String args[]) throws Exception
{
System.out.println("Server started at port:"+port);
DatagramSocket serverSocket = new DatagramSocket(port);
/**
* Formula for lag = (byte_size/sample_rate)*2
* Byte size 9728 will produce ~ 0.45 seconds of lag. Voice slightly broken.
* Byte size 1400 will produce ~ 0.06 seconds of lag. Voice extremely broken.
* Byte size 4000 will produce ~ 0.18 seconds of lag. Voice slightly more broken then 9728.
*/
byte[] receiveData = new byte[4096];
format = new AudioFormat(sampleRate, 16, 1, true, false);
dataLineInfo = new DataLine.Info(SourceDataLine.class, format);
sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceDataLine.open(format);
sourceDataLine.start();
//FloatControl volumeControl = (FloatControl) sourceDataLine.getControl(FloatControl.Type.MASTER_GAIN);
//volumeControl.setValue(1.00f);
DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);
ByteArrayInputStream baiss = new ByteArrayInputStream(receivePacket.getData());
while (status == true)
{
serverSocket.receive(receivePacket);
ais = new AudioInputStream(baiss, format, receivePacket.getLength());
toSpeaker(receivePacket.getData());
}
sourceDataLine.drain();
sourceDataLine.close();
}
public static void toSpeaker(byte soundbytes[]) {
try
{
System.out.println("At the speaker");
sourceDataLine.write(soundbytes, 0, soundbytes.length);
} catch (Exception e) {
System.out.println("Not working in speakers...");
e.printStackTrace();
}
}
}
答案 0 :(得分：6)
所以，我用正弦波填充麦克风（或某些模糊的东西，类似于正弦波），你的程序运行正常。
我的具体变化是：
package audioclient;
import java.io.*;
import java.net.*;
import java.nio.ByteBuffer;
import javax.sound.sampled.*;
public class Mic {
public byte[] buffer;
private int port;
static AudioInputStream ais;
public static void main(String[] args) {
TargetDataLine line;
DatagramPacket dgp;
AudioFormat.Encoding encoding = AudioFormat.Encoding.PCM_SIGNED;
float rate = 44100.0f;
int channels = 2;
int sampleSize = 16;
boolean bigEndian = true;
InetAddress addr;
AudioFormat format = new AudioFormat(encoding, rate, sampleSize, channels, (sampleSize / 8) * channels, rate, bigEndian);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
if (!AudioSystem.isLineSupported(info)) {
System.out.println("Line matching " + info + " not supported.");
return;
}
try {
line = (TargetDataLine) AudioSystem.getLine(info);
//TOTALLY missed this.
int buffsize = line.getBufferSize() / 5;
buffsize += 512;
line.open(format);
line.start();
int numBytesRead;
byte[] data = new byte[buffsize];
/*
* MICK's injection: We have a buffsize of 512; it is best if the frequency
* evenly fits into this (avoid skips, bumps, and pops). Additionally, 44100 Hz,
* with two channels and two bytes per sample. That's four bytes; divide
* 512 by it, you have 128.
*
* 128 samples, 44100 per second; that's a minimum of 344 samples, or 172 Hz.
* Well within hearing range; slight skip from the uneven division. Maybe
* bump it up to 689 Hz.
*
* That's a sine wave of shorts, repeated twice for two channels, with a
* wavelength of 32 samples.
*
* Note: Changed my mind, ignore specific numbers above.
*
*/
{
final int λ = 16;
ByteBuffer buffer = ByteBuffer.allocate(λ * 2 * 8);
for(int j = 0; j < 2; j++) {
for(double i = 0.0; i < λ; i++) {
System.out.println(j + " " + i);
//once for each sample
buffer.putShort((short)(Math.sin(Math.PI * (λ/i)) * Short.MAX_VALUE));
buffer.putShort((short)(Math.sin(Math.PI * (λ/i)) * Short.MAX_VALUE));
}
}
data = buffer.array();
}
addr = InetAddress.getByName("127.0.0.1");
try(DatagramSocket socket = new DatagramSocket()) {
while (true) {
for(byte b : data) System.out.print(b + " ");
// Read the next chunk of data from the TargetDataLine.
// numBytesRead = line.read(data, 0, data.length);
for(int i = 0; i < 64; i++) {
byte b = data[i];
System.out.print(b + " ");
}
System.out.println();
// Save this chunk of data.
dgp = new DatagramPacket(data, data.length, addr, 50005);
for(int i = 0; i < 64; i++) {
byte b = dgp.getData()[i];
System.out.print(b + " ");
}
System.out.println();
socket.send(dgp);
}
}
} catch (LineUnavailableException e) {
e.printStackTrace();
} catch (UnknownHostException e) {
// TODO: handle exception
} catch (SocketException e) {
// TODO: handle exception
} catch (IOException e2) {
// TODO: handle exception
}
}
}
显然我把它误解为一个512字节长的部分并且使正弦波拙劣，但问题是，它产生的声音恰好是它的意思 - 在一个特定的音高上发出令人头疼的嘎嘎声。
考虑到这一点，我并不怀疑代码中明确存在问题。我要检查的第一件事是你的系统正在为音频点击哪一行。你有多个麦克风连接？可能是网络摄像头麦克风？您可以使用PulseAudio Volume Control等实用程序来检查。如果您还没有检查过麦克风的功能，那么您也可以这样做;他们确实有生命。
对音频流中的比特进行加扰并不常见，也不困难;但我没有看到你可以做到的任何地方。
一种想法可能是修改程序以尝试在本地播放声音，然后再将其发送到服务器。这样，您至少可以确定问题是在麦克风之前还是之后。
答案 1 :(得分：2)
当客户端和服务器使用不同大小的数据缓冲区时，将会被截断，并可能导致其中一个或两个产生伪像。
您的服务器缓冲区大小设置为byte[] receiveData = new byte[4096];
您的客户端缓冲区大小由于某种原因是动态的，并设置为byte[] data = new byte[buffsize];
将客户端缓冲区大小设置为静态4096以匹配服务器：byte[] data = new byte[4096];
或者只是确保它们的尺寸相同......
答案 2 :(得分：1)
所以这是一个老问题，但是解决这个问题对我有所帮助，我想我发现的结果可能对其他人有帮助。.这就是我解决您描述的问题的方式：
在我的机器上进行更改
boolean bigEndian = true;
到
boolean bigEndian = false;
解决了白噪声问题（显然是字节顺序问题）
如果这是您所做的唯一更改，则结果音频将具有较低的音调，这是由于以下事实：在麦克风侧您收集了2个通道，而在扬声器侧您通过一个通道播放了。 / p>
要解决此问题，只需更改此行：
format = new AudioFormat(sampleRate, 16, 1, true, false);
到
format = new AudioFormat(sampleRate, 16, 2, true, false);
然后音频应该清晰易懂
答案 3 :(得分：0)
我建议您首先在文件中写入客户端上录制的音频。这将使您能够验证捕获的音频是否正常。您可以使用sox等实用程序将PCM转换为WAV。
答案 4 :(得分：0)
在客户端和服务器上匹配音频格式很重要，例如将Client.java中的音频格式更改为：format = new AudioFormat(sampleRate, 16, 1, true, false);
您还需要在两个程序上使用相同的缓冲区大小。