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java - 在每个 RTP 数据包中捕获 8kHz、16 位线性样本和 4 帧 20ms 音频

转载 作者:塔克拉玛干 更新时间:2023-11-02 19:57:45 27 4
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我需要编写简单的 Java 客户端程序来捕获实时音频流。

要求

  • RTP 音频数据包。
  • 8kHz、16 位线性样本(线性 PCM)。
  • 将在每个 RTP 数据包中发送 4 帧 20ms 音频。

经过一些搜索,我在互联网上找到了示例代码来捕获音频,但它会播放哔声。

代码

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 = 31007;
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);

@SuppressWarnings("resource")
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, 2, 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)
{
System.out.println("Reciving Packets");
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();
}
}
}

我想我找不到合适的格式来捕获以给定格式发送的数据包?

任何人都可以帮助我找到合适的AudioFormat 来捕获此音频流或任何指向相同的链接对我有帮助...谢谢...:)

回答

float sampleRate = 8000;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);

UDP + RTP 数据包格式

enter image description here

同时从数据中缓冲负 12 个字节,因为它包含 RTP header 信息。

receivePacket = new DatagramPacket(receiveData, receiveData.length);
byte[] packet = new byte[receivePacket.getLength() - 12];
serverSocket.receive(receivePacket);
packet = Arrays.copyOfRange(receivePacket.getData(), 12, receivePacket.getLength());

希望这对您将来有所帮助,或者如果不对,请随时纠正谢谢..

最佳答案

你可以试试这个基于数据报套接字的客户端和服务器的实现。它使用单声道 8000Hz 16bit 签名大端音频格式。服务器在端口号 9786 上运行,而客户端使用端口号 8786。我想代码很容易理解。

服务器:

import java.io.*;
import java.net.*;
import javax.sound.sampled.*;

public class Server {

ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;

private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
}

public static void main(String args[]) {
new Server().runVOIP();
}

public void runVOIP() {
try {
DatagramSocket serverSocket = new DatagramSocket(9786);
byte[] receiveData = new byte[4096];
while (true) {
DatagramPacket receivePacket = new DatagramPacket(receiveData, receiveData.length);
serverSocket.receive(receivePacket);
System.out.println("RECEIVED: " + receivePacket.getAddress().getHostAddress() + " " + receivePacket.getPort());
try {
byte audioData[] = receivePacket.getData();
InputStream byteInputStream = new ByteArrayInputStream(audioData);
AudioFormat adFormat = getAudioFormat();
InputStream = new AudioInputStream(byteInputStream, adFormat, audioData.length / adFormat.getFrameSize());
DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, adFormat);
sourceLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
sourceLine.open(adFormat);
sourceLine.start();
Thread playThread = new Thread(new PlayThread());
playThread.start();
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
} catch (Exception e) {
e.printStackTrace();
}
}

class PlayThread extends Thread {

byte tempBuffer[] = new byte[4096];

public void run() {
try {
int cnt;
while ((cnt = InputStream.read(tempBuffer, 0, tempBuffer.length)) != -1) {
if (cnt > 0) {
sourceLine.write(tempBuffer, 0, cnt);
}
}
} catch (Exception e) {
System.out.println(e);
System.exit(0);
}
}
}
}

客户:

import java.io.*;
import java.net.*;
import javax.sound.sampled.*;

public class Client {

boolean stopaudioCapture = false;
ByteArrayOutputStream byteOutputStream;
AudioFormat adFormat;
TargetDataLine targetDataLine;
AudioInputStream InputStream;
SourceDataLine sourceLine;

public static void main(String args[]) {
new Client();
}

public Client() {
captureAudio();
}

private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
}

private void captureAudio() {
try {
adFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(TargetDataLine.class, adFormat);
targetDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
targetDataLine.open(adFormat);
targetDataLine.start();

Thread captureThread = new Thread(new CaptureThread());
captureThread.start();
} catch (Exception e) {
StackTraceElement stackEle[] = e.getStackTrace();
for (StackTraceElement val : stackEle) {
System.out.println(val);
}
System.exit(0);
}
}

class CaptureThread extends Thread {

byte tempBuffer[] = new byte[4096];

@Override
public void run() {
stopaudioCapture = false;
try {
DatagramSocket clientSocket = new DatagramSocket(8786);
InetAddress IPAddress = InetAddress.getByName("127.0.0.1");
int cnt;
while (!stopaudioCapture) {
cnt = targetDataLine.read(tempBuffer, 0, tempBuffer.length);
if (cnt > 0) {
DatagramPacket sendPacket = new DatagramPacket(tempBuffer, tempBuffer.length, IPAddress, 9786);
clientSocket.send(sendPacket);
}
}
} catch (Exception e) {
System.out.println("CaptureThread::run()" + e);
System.exit(0);
}
}
}

}

关于java - 在每个 RTP 数据包中捕获 8kHz、16 位线性样本和 4 帧 20ms 音频,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/32368835/

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