Java 实现Rtsp 转rtmp,hls,flv-Toy模板网

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服务支撑：FFmpeg + srs(流媒体服务器)

整个流程是 java 调用 FFmpeg 收流转码并把流推 rtmp 到流媒体服务，流媒体服务再分发流到公网（如果使用得流媒体服务器支持转协议可以分发 flv ,hls等）

搭建流媒体服务:

1. SRS (Simple Realtime Server) | SRS （本例子使用的是SrS 安装使用docker ）

2.GitHub - ZLMediaKit/ZLMediaKit: WebRTC/RTSP/RTMP/HTTP/HLS/HTTP-FLV/WebSocket-FLV/HTTP-TS/HTTP-fMP4/WebSocket-TS/WebSocket-fMP4/GB28181/SRT server and client framework based on C++11

3.nginx实现（自己百度，他这个只实现了rtmp 要单独下载插件做支持）

4. 其他的还有收费的那种

2.服务器安装FFmpeg yum 可以安装

java 服务实现调用ffmpeg开始拉rtsp 转码推流

1.ProcessManager 用于执行指令以及关闭这个流等操作

package io.renren.common.live;

import cn.hutool.core.thread.ThreadUtil;
import org.springframework.beans.factory.DisposableBean;
import java.io.BufferedReader;
import java.io.IOException;
import java.io.InputStreamReader;
import java.lang.ref.WeakReference;
import java.util.HashMap;
import java.util.Map;

/**
 * @author chenkang
 * @date 2023年8月3日09:43:21
 */
public class ProcessManager implements DisposableBean {

    private Map<String, WeakReference<Process>> processMap=new HashMap<>();

    /**
     * 启动一个进程
     * @param processName 进程名称key
     * @param command 执行指令
     */
    public void startProcess(String processName, String command) {
        ThreadUtil.execAsync(() -> {
            try {
                ProcessBuilder processBuilder = new ProcessBuilder(command.split(" "));
                Process process = processBuilder.start();
                processMap.put(processName, new WeakReference<>(process));
                BufferedReader reader = new BufferedReader(new InputStreamReader(process.getErrorStream()));
                String line;
                while ((line = reader.readLine()) != null) {
                    System.out.println(processName + ": " + line);
                }
                int exitCode = process.waitFor();
                System.out.println(processName + ": Process execution completed with exit code: " + exitCode);
            } catch (IOException | InterruptedException e) {
                e.printStackTrace();
            }
        });
    }

    /**
     *  销毁
     * @param processName key
     */
    public void terminateProcess(String processName) {
        WeakReference<Process> weakRef = processMap.get(processName);
        if (weakRef != null) {
            Process process = weakRef.get();
            if (process != null) {
                process.destroy();
            }
            processMap.remove(processName);
        }
    }



    private void terminateAllProcesses() {
        for (WeakReference<Process> weakRef : processMap.values()) {
            Process process = weakRef.get();
            if (process != null) {
                process.destroy();
            }
        }
    }

    @Override
    public void destroy() throws Exception {
        this.terminateAllProcesses();
        this.processMap.clear();
    }
}

对接的是大华的摄像头

/**
* 开始推流
* 备注：现在客户的设备是NVR NVR 下大概有54个摄像头公网映射rtsp 554 端口根据channel 开区分是那个摄像头
* 必须条件：1.服务端要安装好 ffmpeg 2.要搭建一个流媒体服务器这个使用的是 srs
* 流程：用户端想要查看某个摄像头->查询到设备信息获取到摄像头的channel 这个是提前维护好的
* ->拿到channel走如下方法调用FFmpeg 执行转码推流指令（客户的摄像头是h265）-》rtsp流会被转码重新设定分辨率码率转h264 并把转码流推向流媒体服务器 srs
* ->客户想看的时候就拉取流媒体服务端的rtmp流完成播放
*
* 其他：
* 1.用户再播放的时候要先确定这个摄像头有没有别的人在观看观看了就不在执行了（1.可以调用srs 接口查询流是不是存在这个比较稳妥 2.或者是 processManager 判断是否在推了）
* 2.有时候用户强制关闭客户端无法感知用户不在观看了，这面还要定时的去调用srs 接口查询闲置的流及时的给关闭同时也要把服务的process 给主动关闭不然一直推
* 这个要先去关闭process再调用接口关闭srs服务的流
*
*
* http://127.0.0.1:1985/api/v1/streams 查询服务端所有流
*{
* "code": 0,
* "server": "vid-f1gt8j3",
* "streams": [
* {
* "id": "vid-143p019",
* //streamName
* "name": "3",
* "vhost": "vid-5847096",
* "app": "live",
* "live_ms": 1691039306435,
* //客户端数量这个要注意默认就有1 个客户端是推流
* "clients": 1,
* "frames": 0,
* "send_bytes": 0,
* "recv_bytes": 1068,
* "kbps": {
* "recv_30s": 0,
* "send_30s": 0
* },
* "publish": {
* //是否正在推流有时候服务端流停推了但是还有客户端在看这个流还能查到但是 active 为false
* "active": false,
* "cid": ""
* },
* "video": null,
* "audio": null
* }
* ]
* }
*
*
*
* http://127.0.0.1:1985/api/v1/clients
*
*{
* "code": 0,
* "server": "vid-f1gt8j3",
* "clients": [
* {
* "id": "868249e9",
* "vhost": "vid-5847096",
* "stream": "vid-778ujy0",
* "ip": "172.17.0.1",
* "pageUrl": "",
* "swfUrl": "",
* "tcUrl": "rtmp://127.0.0.1:1935/live",
* "url": "/live/9",
* //类型是 fmle-publish 推流删除掉这个推流就会停止
* //类型是 rtmp-play 拉流删除掉这个拉流就会停止
* //剔除方法 Method DELETE api /api/v1/clients/{id} 停止推流/踢掉用户端
* "type": "fmle-publish",
* "publish": true,
* "alive": 16.18,
* "kbps": {
* "recv_30s": 0,
* "send_30s": 0
* }
* }
* ]
* }
*
*
* @param channel
*/


    @GetMapping("/start")
    @ResponseBody
    public void start(@RequestParam(defaultValue = "1") String channel){
        RtspUrlBuilder builder = new RtspUrlBuilder();
        RtmpUrlBuilder rtmpUrlBuilder = new RtmpUrlBuilder();
        //构建 rtsp 这个是客户的nvr rtsp 地址 只有channel 是灵活的 他们是64路 现在接了50多摄像头对应50 多路channel
        String rstp = builder.setUsername("admin")
                .setPassword("xx")
                .setIpAddress("xx")
                .setChannel(channel)
                .build();
        //这个是流媒体服务器的rtmp 推流地址
        String rtmp = rtmpUrlBuilder.setApplication("live").setStreamName(channel).build();
        final String vcodec="libx264";
        String camera1=String.format(RTSP_RTMP, rstp,vcodec,rtmp);
        //TODO 判断是否已经再推了 推就直接返回拉流地址
        processManager.startProcess(channel,camera1);
        //拉流地址和推流地址是一至的 除非 java 服务和srs 在一台服务器 那么  推流地址 rtmp ip为127.0.0.1 拉流 rtmp ip 为公网
        // 就是java通过ffmpeg 收流转发到本地 rtmp srs分发流 到公网去
        System.out.println("拉流地址："+rtmp);
    }

    @GetMapping("/end")
    @ResponseBody
    public void end(String channel){
        processManager.terminateProcess(channel);
    }

两个辅助类：

package io.renren.common.live;

/**
 * @author chenkang
 * @date 2023-8-3 12:27
 */
public class RtspUrlBuilder {
    private String username;
    private String password;
    private String ipAddress;
    private int port;
    private String channel;
    private int subtype;

    public RtspUrlBuilder() {
        // 默认端口为554
        this.port = 554;
        // 默认子类型为0
        this.subtype = 0;
    }

    public RtspUrlBuilder setUsername(String username) {
        this.username = username;
        return this;
    }

    public RtspUrlBuilder setPassword(String password) {
        this.password = password;
        return this;
    }

    public RtspUrlBuilder setIpAddress(String ipAddress) {
        this.ipAddress = ipAddress;
        return this;
    }

    public RtspUrlBuilder setPort(int port) {
        this.port = port;
        return this;
    }

    public RtspUrlBuilder setChannel(String channel) {
        this.channel = channel;
        return this;
    }

    public RtspUrlBuilder setSubtype(int subtype) {
        this.subtype = subtype;
        return this;
    }

    public String build() {
        return "rtsp://" + username + ":" + password + "@" + ipAddress + ":" + port +
                "/cam/realmonitor?channel=" + channel + "&subtype=" + subtype;
    }
}

package io.renren.common.live;

/**
 * @author chenkang
 * @date 2023-8-3 12:30
 */
public class RtmpUrlBuilder {
    private String ipAddress;
    private int port;
    private String application;
    private String streamName;

    public RtmpUrlBuilder() {
        // 默认IP地址为127.0.0.1
        this.ipAddress = "127.0.0.1";
        // 默认端口为1935
        this.port = 1935;
    }

    public RtmpUrlBuilder setIpAddress(String ipAddress) {
        this.ipAddress = ipAddress;
        return this;
    }

    public RtmpUrlBuilder setPort(int port) {
        this.port = port;
        return this;
    }

    public RtmpUrlBuilder setApplication(String application) {
        this.application = application;
        return this;
    }

    public RtmpUrlBuilder setStreamName(String streamName) {
        this.streamName = streamName;
        return this;
    }

    public String build() {
        return  "rtmp://" + ipAddress + ":" + port + "/" + application + "/" + streamName;
    }
}

访问得话应该是访问流媒体得ip 文章来源地址https://www.toymoban.com/news/detail-645535.html

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