用java实现UDP接收rtp的音频流,并解析rtp音频流将PCMA/PCMU格式音频数据并转换成PCM格式,如何比特率不是16要调整16比特,并给出调用案例和详细的注释
时间: 2023-12-17 15:05:45 浏览: 162
实现UDP接收rtp的音频流,可以使用Java的DatagramSocket类来进行网络数据包的接收,解析rtp音频流可以参考RFC3550中的相关规范。将PCMA/PCMU格式音频数据转换成PCM格式,可以使用Java的AudioFormat类和AudioInputStream类来进行音频格式的转换。具体实现的代码如下:
```java
import java.io.ByteArrayInputStream;
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.util.Arrays;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.TargetDataLine;
public class RTPAudioReceiver {
private static final int RTP_HEADER_SIZE = 12;
private static final int UDP_PACKET_SIZE = 1500;
private static final int AUDIO_SAMPLE_RATE = 8000;
private static final int AUDIO_SAMPLE_SIZE_IN_BITS = 16;
private static final int AUDIO_CHANNELS = 1;
private static final boolean AUDIO_SIGNED = true;
private static final boolean AUDIO_BIG_ENDIAN = true;
private DatagramSocket socket;
private InetAddress remoteAddress;
private int remotePort;
private AudioFormat audioFormat;
private DataLine.Info dataLineInfo;
private TargetDataLine targetDataLine;
public RTPAudioReceiver(String remoteIP, int remotePort) throws IOException {
this.remoteAddress = InetAddress.getByName(remoteIP);
this.remotePort = remotePort;
this.socket = new DatagramSocket();
this.audioFormat = new AudioFormat(
AUDIO_SAMPLE_RATE, AUDIO_SAMPLE_SIZE_IN_BITS,
AUDIO_CHANNELS, AUDIO_SIGNED, AUDIO_BIG_ENDIAN);
this.dataLineInfo = new DataLine.Info(TargetDataLine.class, audioFormat);
this.targetDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
}
public void start() throws Exception {
targetDataLine.open(audioFormat);
targetDataLine.start();
byte[] buffer = new byte[UDP_PACKET_SIZE];
while (true) {
DatagramPacket packet = new DatagramPacket(buffer, buffer.length, remoteAddress, remotePort);
socket.receive(packet);
byte[] rtpPacket = Arrays.copyOf(packet.getData(), packet.getLength());
byte[] pcmData = decodeRTP(rtpPacket);
if (pcmData != null) {
playAudio(pcmData);
}
}
}
private byte[] decodeRTP(byte[] rtpPacket) {
int payloadType = (rtpPacket[1] & 0xFF) & 0x7F;
if (payloadType != 0) {
// 不支持的payload type
return null;
}
int sequenceNumber = ((rtpPacket[2] & 0xFF) << 8) | (rtpPacket[3] & 0xFF);
int timestamp = ((rtpPacket[4] & 0xFF) << 24) | ((rtpPacket[5] & 0xFF) << 16) | ((rtpPacket[6] & 0xFF) << 8) | (rtpPacket[7] & 0xFF);
int ssrc = ((rtpPacket[8] & 0xFF) << 24) | ((rtpPacket[9] & 0xFF) << 16) | ((rtpPacket[10] & 0xFF) << 8) | (rtpPacket[11] & 0xFF);
byte[] payload = Arrays.copyOfRange(rtpPacket, RTP_HEADER_SIZE, rtpPacket.length);
if (payload.length == 0) {
return null;
}
byte[] pcmData = decodePayload(payload);
return pcmData;
}
private byte[] decodePayload(byte[] payload) {
int numSamples = payload.length;
byte[] pcmData = new byte[numSamples * 2];
for (int i = 0; i < numSamples; i++) {
int pcmValue = ((int) payload[i]) << 8;
pcmData[i * 2] = (byte) (pcmValue & 0xFF);
pcmData[i * 2 + 1] = (byte) ((pcmValue >> 8) & 0xFF);
}
return pcmData;
}
private void playAudio(byte[] pcmData) {
AudioInputStream audioInputStream = new AudioInputStream(
new ByteArrayInputStream(pcmData), audioFormat, pcmData.length / audioFormat.getFrameSize());
ByteArrayOutputStream baos = new ByteArrayOutputStream();
try {
AudioSystem.write(audioInputStream, AudioFileFormat.Type.WAVE, baos);
} catch (IOException e) {
e.printStackTrace();
}
byte[] audioBytes = baos.toByteArray();
try {
baos.close();
audioInputStream.close();
} catch (IOException e) {
e.printStackTrace();
}
targetDataLine.write(audioBytes, 0, audioBytes.length);
}
public static void main(String[] args) throws Exception {
RTPAudioReceiver receiver = new RTPAudioReceiver("127.0.0.1", 12345);
receiver.start();
}
}
```
调用示例:
```java
RTPAudioReceiver receiver = new RTPAudioReceiver("127.0.0.1", 12345);
receiver.start();
```
其中,参数"127.0.0.1"表示远程IP地址,参数12345表示远程端口号。
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