java实现g711a打包发送rtp(一)

参考文章:https://blog.youkuaiyun.com/qq_38795209/article/details/113942389?login=from_csdn

        摄像头语音对讲/广播实现,需要发送g711a的RTP包,参考网上资料 , 示例代码如下,已验证基于GB28181语音广播流程,最后通过该工具类发送语音数据,华为摄像头可进行语音播放

RtpPacket.java

package com.genersoft.iot.vmp.vmanager.gb28181.talk.packet;

import java.io.FileInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.net.*;

/**
 * 参考出处:https://blog.youkuaiyun.com/qq_38795209/article/details/113942389?login=from_csdn
 * g711a a-law rtp 数据包
 * 如何计算打包发送间隔、打包字节数:
 * 音频的帧率 				fps = 20
 * 采样率        			sample_rate = 8000 HZ
 * 码率			 		    bitrate = 64000	bps
 * 打包发送间隔   		 	send_interval = 1 / 20 = 0.05s = 50000us
 * 每次打包需要音频数据长度     audio_need_len = 64000 bps * 0.05s = 3200 bit = 400 bytes    码率/帧率/8
 * udp发送数据长度			send_len =  （rtp包头12 bytes ）+ audio_need_len
 */
public class RtpPacket {

    public static int RTP_PAYLOAD_TYPE_PCMU = 0; // u-law
    public static int RTP_PAYLOAD_TYPE_PCMA = 8; // a-law

    private byte version = 2;

    private byte padding = 0;

    private byte extension = 0;

    private byte csrc_count = 0;

    private byte marker = 1;

    private byte playload_type = 0;

    private int sequence_number = 0;

    private long timestamp = 0;

    private long ssrc = 0;


    public RtpPacket(long ssrcVal) {
        ssrc = ssrcVal;
    }

    public byte[] packet_g711a(byte[] audioData, int audioLen) {

        int    rtp_head_len = 12;
        byte[] payload      = new byte[rtp_head_len + audioLen];

        playload_type = (byte) RTP_PAYLOAD_TYPE_PCMA; // g711a a-law
        sequence_number++;
        timestamp += audioLen;

        // 12位 RTP 包头
        payload[0] = (byte) ((version << 6) + (padding << 5) + (extension << 4) + csrc_count);
        payload[1] = (byte) ((marker << 7) + playload_type);
        payload[2] = (byte) (sequence_number / (0xff + 1));
        payload[3] = (byte) (sequence_number % (0xff + 1));
        payload[4] = (byte) ((timestamp / (0xffff + 1)) / (0xff + 1));
        payload[5] = (byte) ((timestamp / (0xffff + 1)) % (0xff + 1));
        payload[6] = (byte) ((timestamp % (0xffff + 1)) / (0xff + 1));
        payload[7] = (byte) ((timestamp % (0xffff + 1)) % (0xff + 1));
        payload[8] = (byte) ((ssrc / (0xffff + 1)) / (0xff + 1));
        payload[9] = (byte) ((ssrc / (0xffff + 1)) % (0xff + 1));
        payload[10] = (byte) ((ssrc % (0xffff + 1)) / (0xff + 1));
        payload[11] = (byte) ((ssrc % (0xffff + 1)) % (0xff + 1));

        System.arraycopy(audioData, 0, payload, rtp_head_len, audioLen);
        return payload;
    }

    public static void main(String[] args) throws IOException {
        long           ssrc             = 255;
        int            send_interval_ms = 50;
        int            audio_need_len   = 400;
        int            localPort        = 56200;
        int            peerPort         = 15062;
        String         peerIP           = "192.168.1.101";
        DatagramSocket ds               = null;
        InetAddress    peerAddress      = null;
        InputStream    inputStream      = new FileInputStream("./mq.g711a");

        try {
            ds = new DatagramSocket(localPort);
        } catch (SocketException e) {
            e.printStackTrace();
            System.exit(1);
        }

        try {
            peerAddress = InetAddress.getByName(peerIP);
        } catch (UnknownHostException e) {
            e.printStackTrace();
            System.exit(1);
        }

        RtpPacket rtp        = new RtpPacket(ssrc);
        byte[]    audio_data = new byte[audio_need_len];

        while (inputStream.read(audio_data) == audio_need_len) {

            byte[]         payload = rtp.packet_g711a(audio_data, audio_data.length);
            DatagramPacket dp      = new DatagramPacket(payload, payload.length, peerAddress, peerPort);

            ds.send(dp);
            //System.out.println(Arrays.toString(payload));
            try {
                Thread.sleep(send_interval_ms);

            } catch (InterruptedException e) {
                e.printStackTrace();
                break;
            }
        }
        ds.close();
    }
}

工具类

package com.genersoft.iot.vmp.vmanager.gb28181.talk.packet;

import org.apache.commons.compress.utils.IOUtils;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;

import java.io.FileInputStream;
import java.io.InputStream;
import java.net.DatagramPacket;
import java.net.DatagramSocket;
import java.net.InetAddress;

/**
 * 发送rtp数据包
 */
public class SendRTPUtil {
    private static final Logger log = LoggerFactory.getLogger(SendRTPUtil.class);

    /**
     * 将指定文件打包成rtp下发到摄像机,注意文件格式 g711a a-law
     *
     * @param localPort  本机rtp服务端口
     * @param cameraIp
     * @param cameraPort
     * @param ssrc
     * @param audioFile
     */
    public static void send(int localPort, String cameraIp, int cameraPort, Long ssrc, String audioFile) {
        DatagramSocket ds          = null;
        InputStream    audioStream = null;

        int send_interval_ms = 50; // 50ms一帧, 帧率 20 fps
        int rtpPacketSize    = 400; // raw data size = 码率/帧率/8
        ssrc = ssrc == null ? 255 : ssrc;
        try {
            ds = new DatagramSocket(localPort);
            InetAddress peerAddress = InetAddress.getByName(cameraIp);

            audioStream = new FileInputStream(audioFile);
            RtpPacket rtpPacket = new RtpPacket(ssrc);

            byte[] audio_data = new byte[rtpPacketSize];
            log.info(">>>>>开始发送rtp: localPort={},peerAddress={},peerPort={},ssrc={},audioFile={}", localPort, cameraIp, cameraPort, ssrc, audioFile);
            int i = 0;
            while (audioStream.read(audio_data) == rtpPacketSize) {

                byte[]         payload = rtpPacket.packet_g711a(audio_data, audio_data.length);
                DatagramPacket dp      = new DatagramPacket(payload, payload.length, peerAddress, cameraPort);
                ds.send(dp);
                log.debug(">>>>>>>发送rtp包:{}",i);
                try {
                    Thread.sleep(send_interval_ms);
                } catch (InterruptedException e) {
                    e.printStackTrace();
                    break;
                }
            }
            log.info(">>>>>发送完成: localPort={},peerAddress={},peerPort={},ssrc={},audioFile={}", localPort, cameraIp, cameraPort, ssrc, audioFile);
        } catch (Exception e) {
            e.printStackTrace();
            log.error(">>>>>发送rtp异常: localPort={},peerAddress={},peerPort={},ssrc={},audioFile={},{}", localPort, cameraIp, cameraPort, ssrc, audioFile, e.getMessage());
        } finally {
            if (ds != null) {
                log.info(">>>>>释放本地端口:{}",ds.getLocalPort());
                ds.disconnect();
                ds.close();
            }

            IOUtils.closeQuietly(audioStream);
        }
    }
}