实时音频流java [英] Live audio stream java
问题描述
我正在另一台 PC 上实现从 MIC 到 Java 服务器的实时流式传输.但我只听到白噪音.
I am implementing live streaming from MIC to java server at another PC. But I am only hearing a white noise.
我已经附加了客户端和服务器程序
I have attached both client and server program
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.与android客户端AudioRecord完美运行.
and the server side is no issue. It is running perfectly with android client AudioRecord.
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();
}
}
}
推荐答案
所以，我用正弦波(或某种模糊的意义上类似于正弦波的东西)填充麦克风，并且您的程序运行良好.
So, I filled the microphone with a sine wave (or something which, in some vague sense, resembles a sine wave), and your program works fine.
我的具体更改如下:
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
}
}
}
很明显，我将它误解为 512 字节长的片段并弄坏了正弦波，但事实是，它产生的声音正是它想要的——在特定音高下发出令人麻木的嘎嘎声.
Obviously I misinterpreted it as a 512-byte-long piece and botched the sine wave, but the thing is, it produced exactly the sound that it was meant to--a mind-numbing rattle at a specific pitch.
考虑到这一点，我不怀疑问题出在您的代码中.我要检查的第一件事是您的系统正在窃听哪条线路以获取音频.您是否连接了多个麦克风?也许是网络摄像头麦克风?您可能会使用 PulseAudio Volume Control 之类的实用程序进行检查.如果您还没有检查过麦克风的功能，您也可以这样做；它们确实有使用寿命.
This in mind, I don't suspect that the problem is explicitly in your code. The first thing I would check is which line your system is tapping for audio. Do you have multiple microphones hooked up? A webcam mic, maybe? You might grab a utility like PulseAudio Volume Control to check. If you haven't already checked on the functionality of your microphone, you might do that too; they do have a lifespan on them.
对音频流中的比特进行加扰并不少见，也并不困难；但我看不到任何地方可以这样做.
It isn't uncommon at all to scramble the bits in an audio stream, nor is it difficult; but I don't see anywhere where you could be doing that.
一个想法可能是修改您的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前还是麦克风后.
One thought might be to modify your program to attempt to play the sound locally, before sending it over to the server. That way, you can at least determine if the problem is pre- or post-Mic.
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