技术标签: RTSP专栏
对于RTP OVER UDP 的实现,我们使用TCP连接来发送RTSP交互,然后创建新的UDP套接字来发送RTP包,和建新的UDP套接字来发送RTCP包。
对于RTP OVER RTSP(TCP)来说,我们会复用使用原先发送RTSP的socket来发送RTP包和RTCP包。
如上面所说,我们复用发送RTSP交互的socket来发送RTP包和RTCP信息,那么对于客户端来说,如何区分这三种数据呢?
我们将这三个分为两类,一类是RTSP,一类是RTP、RTCP
发送RTSP信息的情况没有变化,还是更以前一样的方式
发送RTP、RTCP包,在每个包前面都加上四个字节
由此我们可知,第一个字节’$'用于与RTSP区分,第二个字节用于区分RTP和RTCP
RTP和RTCP的channel是在RTSP的SETUP过程中,客户端发送给服务端的
所以现在RTP的打包方式要在之前的每个RTP包前面加上四个字节,如下所示
经过上面的介绍,我们知道RTP OVER TCP和RTP OVER UDP的RTP发包方式是不同的,RTP OVER TCP需要在整一个RTP包前面加上四个字节,为此我修改了RTP发包部分
struct RtpPacket
{
char header[4];
struct RtpHeader rtpHeader;
uint8_t payload[0];
};
header:前四个字节
rtpHeader:RTP包头部
payload:RTP包载荷
RTP的发包函数修改
每次发包前都需要添加四个字节的头,并且通过tcp发送
rtpSendPacket()
{
...
rtpPacket->header[0] = '$';
rtpPacket->header[1] = rtpChannel;
rtpPacket->header[2] = (size & 0xFF00 ) >> 8;
rtpPacket->header[3] = size & 0xFF;
...
send(...);
...
}
下面开始介绍RTP OVER TCP服务器的实现过程
main()
{
serverSockfd = createTcpSocket();
bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
listen(serverSockfd, 10);
...
while(1)
{
...
}
}
main()
{
...
while(1)
{
acceptClient(serverSockfd, clientIp, &clientPort);
doClient(clientSockfd, clientIp, clientPort);
}
}
接收客户端连接后,执行doClient处理客户端请求
接收请求后,解析请求,先解析方法,再解析序列号,如果是SETUP,那么就将RTP通道和RTCP通道解析出来
然后处理不同的请求方法
doClient()
{
while(1)
{
/* 接收数据 */
recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
/* 解析命令 */
sscanf(line, "%s %s %s\r\n", method, url, version);
...
sscanf(line, "CSeq: %d\r\n", &cseq);
...
if(!strcmp(method, "SETUP"))
sscanf(line, "Transport: RTP/AVP/TCP;unicast;interleaved=%hhu-%hhu\r\n",
&rtpChannel, &rtcpChannel);
/* 处理请求 */
if(!strcmp(method, "OPTIONS"))
handleCmd_OPTIONS(sBuf, cseq)
else if(!strcmp(method, "DESCRIBE"))
handleCmd_DESCRIBE(sBuf, cseq, url);
else if(!strcmp(method, "SETUP"))
handleCmd_SETUP(sBuf, cseq, rtpChannel);
else if(!strcmp(method, "PLAY"))
handleCmd_PLAY(sBuf, cseq);
send(clientSockfd, sBuf, strlen(sBuf), 0);
}
}
OPTIONS
handleCmd_OPTIONS()
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n"
"\r\n",
cseq);
}
DESCRIBE
handleCmd_DESCRIBE()
{
sprintf(sdp, "v=0\r\n"
"o=- 9%ld 1 IN IP4 %s\r\n"
"t=0 0\r\n"
"a=control:*\r\n"
"m=video 0 RTP/AVP 96\r\n"
"a=rtpmap:96 H264/90000\r\n"
"a=control:track0\r\n",
time(NULL), localIp);
sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n"
"Content-Base: %s\r\n"
"Content-type: application/sdp\r\n"
"Content-length: %d\r\n\r\n"
"%s",
cseq,
url,
strlen(sdp),
sdp);
}
SETUP
handleCmd_SETUP()
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Transport: RTP/AVP/TCP;unicast;interleaved=%hhu-%hhu\r\n"
"Session: 66334873\r\n"
"\r\n",
cseq,
rtpChannel,
rtpChannel+1
);
}
PLAY
handleCmd_PLAY()
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Range: npt=0.000-\r\n"
"Session: 66334873; timeout=60\r\n\r\n",
cseq);
}
在发送完PLAY回复之后,开始发送RTP包
doClient()
{
...
while(1)
{
...
send(clientSockfd, sBuf, strlen(sBuf), 0);
if(!strcmp(method, "PLAY"))
{
while(1)
{
/* 获取一帧数据 */
getFrameFromH264File(fd, frame, 500000);
/* 发送RTP包 */
rtpSendH264Frame(clientSockfd);
}
}
}
}
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <assert.h>
#include "tcp_rtp.h"
#define H264_FILE_NAME "test.h264"
#define SERVER_PORT 8554
#define BUF_MAX_SIZE (1024*1024)
static int createTcpSocket()
{
int sockfd;
int on = 1;
sockfd = socket(AF_INET, SOCK_STREAM, 0);
if(sockfd < 0)
return -1;
setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
return sockfd;
}
static int createUdpSocket()
{
int sockfd;
int on = 1;
sockfd = socket(AF_INET, SOCK_DGRAM, 0);
if(sockfd < 0)
return -1;
setsockopt(sockfd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on));
return sockfd;
}
static int bindSocketAddr(int sockfd, const char* ip, int port)
{
struct sockaddr_in addr;
addr.sin_family = AF_INET;
addr.sin_port = htons(port);
addr.sin_addr.s_addr = inet_addr(ip);
if(bind(sockfd, (struct sockaddr *)&addr, sizeof(struct sockaddr)) < 0)
return -1;
return 0;
}
static int acceptClient(int sockfd, char* ip, int* port)
{
int clientfd;
socklen_t len = 0;
struct sockaddr_in addr;
memset(&addr, 0, sizeof(addr));
len = sizeof(addr);
clientfd = accept(sockfd, (struct sockaddr *)&addr, &len);
if(clientfd < 0)
return -1;
strcpy(ip, inet_ntoa(addr.sin_addr));
*port = ntohs(addr.sin_port);
return clientfd;
}
static inline int startCode3(char* buf)
{
if(buf[0] == 0 && buf[1] == 0 && buf[2] == 1)
return 1;
else
return 0;
}
static inline int startCode4(char* buf)
{
if(buf[0] == 0 && buf[1] == 0 && buf[2] == 0 && buf[3] == 1)
return 1;
else
return 0;
}
static char* findNextStartCode(char* buf, int len)
{
int i;
if(len < 3)
return NULL;
for(i = 0; i < len-3; ++i)
{
if(startCode3(buf) || startCode4(buf))
return buf;
++buf;
}
if(startCode3(buf))
return buf;
return NULL;
}
static int getFrameFromH264File(int fd, char* frame, int size)
{
int rSize, frameSize;
char* nextStartCode;
if(fd < 0)
return fd;
rSize = read(fd, frame, size);
if(!startCode3(frame) && !startCode4(frame))
return -1;
nextStartCode = findNextStartCode(frame+3, rSize-3);
if(!nextStartCode)
{
//lseek(fd, 0, SEEK_SET);
//frameSize = rSize;
return -1;
}
else
{
frameSize = (nextStartCode-frame);
lseek(fd, frameSize-rSize, SEEK_CUR);
}
return frameSize;
}
static int rtpSendH264Frame(int socket, int rtpChannel, struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize)
{
uint8_t naluType; // nalu第一个字节
int sendBytes = 0;
int ret;
naluType = frame[0];
if (frameSize <= RTP_MAX_PKT_SIZE) // nalu长度小于最大包场:单一NALU单元模式
{
/*
* 0 1 2 3 4 5 6 7 8 9
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |F|NRI| Type | a single NAL unit ... |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
memcpy(rtpPacket->payload, frame, frameSize);
ret = rtpSendPacket(socket, rtpChannel, rtpPacket, frameSize);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
if ((naluType & 0x1F) == 7 || (naluType & 0x1F) == 8) // 如果是SPS、PPS就不需要加时间戳
goto out;
}
else // nalu长度小于最大包场:分片模式
{
/*
* 0 1 2
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | FU indicator | FU header | FU payload ... |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/*
* FU Indicator
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F|NRI| Type |
* +---------------+
*/
/*
* FU Header
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |S|E|R| Type |
* +---------------+
*/
int pktNum = frameSize / RTP_MAX_PKT_SIZE; // 有几个完整的包
int remainPktSize = frameSize % RTP_MAX_PKT_SIZE; // 剩余不完整包的大小
int i, pos = 1;
/* 发送完整的包 */
for (i = 0; i < pktNum; i++)
{
rtpPacket->payload[0] = (naluType & 0x60) | 28;
rtpPacket->payload[1] = naluType & 0x1F;
if (i == 0) //第一包数据
rtpPacket->payload[1] |= 0x80; // start
else if (remainPktSize == 0 && i == pktNum - 1) //最后一包数据
rtpPacket->payload[1] |= 0x40; // end
memcpy(rtpPacket->payload+2, frame+pos, RTP_MAX_PKT_SIZE);
ret = rtpSendPacket(socket, rtpChannel, rtpPacket, RTP_MAX_PKT_SIZE+2);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
pos += RTP_MAX_PKT_SIZE;
}
/* 发送剩余的数据 */
if (remainPktSize > 0)
{
rtpPacket->payload[0] = (naluType & 0x60) | 28;
rtpPacket->payload[1] = naluType & 0x1F;
rtpPacket->payload[1] |= 0x40; //end
memcpy(rtpPacket->payload+2, frame+pos, remainPktSize+2);
ret = rtpSendPacket(socket, rtpChannel, rtpPacket, remainPktSize+2);
if(ret < 0)
return -1;
rtpPacket->rtpHeader.seq++;
sendBytes += ret;
}
}
out:
return sendBytes;
}
static char* getLineFromBuf(char* buf, char* line)
{
while(*buf != '\n')
{
*line = *buf;
line++;
buf++;
}
*line = '\n';
++line;
*line = '\0';
++buf;
return buf;
}
static int handleCmd_OPTIONS(char* result, int cseq)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Public: OPTIONS, DESCRIBE, SETUP, PLAY\r\n"
"\r\n",
cseq);
return 0;
}
static int handleCmd_DESCRIBE(char* result, int cseq, char* url)
{
char sdp[500];
char localIp[100];
sscanf(url, "rtsp://%[^:]:", localIp);
sprintf(sdp, "v=0\r\n"
"o=- 9%ld 1 IN IP4 %s\r\n"
"t=0 0\r\n"
"a=control:*\r\n"
"m=video 0 RTP/AVP 96\r\n"
"a=rtpmap:96 H264/90000\r\n"
"a=control:track0\r\n",
time(NULL), localIp);
sprintf(result, "RTSP/1.0 200 OK\r\nCSeq: %d\r\n"
"Content-Base: %s\r\n"
"Content-type: application/sdp\r\n"
"Content-length: %d\r\n\r\n"
"%s",
cseq,
url,
strlen(sdp),
sdp);
return 0;
}
static int handleCmd_SETUP(char* result, int cseq, uint8_t rtpChannel)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Transport: RTP/AVP/TCP;unicast;interleaved=%hhu-%hhu\r\n"
"Session: 66334873\r\n"
"\r\n",
cseq,
rtpChannel,
rtpChannel+1
);
return 0;
}
static int handleCmd_PLAY(char* result, int cseq)
{
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Range: npt=0.000-\r\n"
"Session: 66334873; timeout=60\r\n\r\n",
cseq);
return 0;
}
static void doClient(int clientSockfd, const char* clientIP, int clientPort)
{
char method[40];
char url[100];
char version[40];
int cseq;
char *bufPtr;
char* rBuf = malloc(BUF_MAX_SIZE);
char* sBuf = malloc(BUF_MAX_SIZE);
char line[400];
uint8_t rtpChannel;
uint8_t rtcpChannel;
while(1)
{
int recvLen;
recvLen = recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
if(recvLen <= 0)
goto out;
rBuf[recvLen] = '\0';
printf("---------------C->S--------------\n");
printf("%s", rBuf);
/* 解析方法 */
bufPtr = getLineFromBuf(rBuf, line);
if(sscanf(line, "%s %s %s\r\n", method, url, version) != 3)
{
printf("parse err\n");
goto out;
}
/* 解析序列号 */
bufPtr = getLineFromBuf(bufPtr, line);
if(sscanf(line, "CSeq: %d\r\n", &cseq) != 1)
{
printf("parse err\n");
goto out;
}
/* 如果是SETUP,那么就再解析channel */
if(!strcmp(method, "SETUP"))
{
while(1)
{
bufPtr = getLineFromBuf(bufPtr, line);
if(!strncmp(line, "Transport:", strlen("Transport:")))
{
sscanf(line, "Transport: RTP/AVP/TCP;unicast;interleaved=%hhu-%hhu\r\n",
&rtpChannel, &rtcpChannel);
break;
}
}
}
if(!strcmp(method, "OPTIONS"))
{
if(handleCmd_OPTIONS(sBuf, cseq))
{
printf("failed to handle options\n");
goto out;
}
}
else if(!strcmp(method, "DESCRIBE"))
{
if(handleCmd_DESCRIBE(sBuf, cseq, url))
{
printf("failed to handle describe\n");
goto out;
}
}
else if(!strcmp(method, "SETUP"))
{
if(handleCmd_SETUP(sBuf, cseq, rtpChannel))
{
printf("failed to handle setup\n");
goto out;
}
}
else if(!strcmp(method, "PLAY"))
{
if(handleCmd_PLAY(sBuf, cseq))
{
printf("failed to handle play\n");
goto out;
}
}
else
{
goto out;
}
printf("---------------S->C--------------\n");
printf("%s", sBuf);
send(clientSockfd, sBuf, strlen(sBuf), 0);
/* 开始播放,发送RTP包 */
if(!strcmp(method, "PLAY"))
{
int frameSize, startCode;
char* frame = malloc(500000);
struct RtpPacket* rtpPacket = (struct RtpPacket*)malloc(500000);
int fd = open(H264_FILE_NAME, O_RDONLY);
assert(fd > 0);
rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_H264, 0,
0, 0, 0x88923423);
printf("start play\n");
while (1)
{
frameSize = getFrameFromH264File(fd, frame, 500000);
if(frameSize < 0)
{
break;
}
if(startCode3(frame))
startCode = 3;
else
startCode = 4;
frameSize -= startCode;
rtpSendH264Frame(clientSockfd, rtpChannel, rtpPacket, frame+startCode, frameSize);
rtpPacket->rtpHeader.timestamp += 90000/25;
usleep(1000*1000/25);
}
free(frame);
free(rtpPacket);
goto out;
}
}
out:
printf("finish\n");
close(clientSockfd);
free(rBuf);
free(sBuf);
}
int main(int argc, char* argv[])
{
int serverSockfd;
int ret;
serverSockfd = createTcpSocket();
if(serverSockfd < 0)
{
printf("failed to create tcp socket\n");
return -1;
}
ret = bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
if(ret < 0)
{
printf("failed to bind addr\n");
return -1;
}
ret = listen(serverSockfd, 10);
if(ret < 0)
{
printf("failed to listen\n");
return -1;
}
printf("rtsp://127.0.0.1:%d\n", SERVER_PORT);
while(1)
{
int clientSockfd;
char clientIp[40];
int clientPort;
clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
if(clientSockfd < 0)
{
printf("failed to accept client\n");
return -1;
}
printf("accept client;client ip:%s,client port:%d\n", clientIp, clientPort);
doClient(clientSockfd, clientIp, clientPort);
}
return 0;
}
tcp_rtp.h
#ifndef _RTP_H_
#define _RTP_H_
#include <stdint.h>
#define RTP_VESION 2
#define RTP_PAYLOAD_TYPE_H264 96
#define RTP_PAYLOAD_TYPE_AAC 97
#define RTP_HEADER_SIZE 12
#define RTP_MAX_PKT_SIZE 1400
/*
*
* 0 1 2 3
* 7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0|7 6 5 4 3 2 1 0
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |V=2|P|X| CC |M| PT | sequence number |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | timestamp |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | synchronization source (SSRC) identifier |
* +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
* | contributing source (CSRC) identifiers |
* : .... :
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
*/
struct RtpHeader
{
/* byte 0 */
uint8_t csrcLen:4;
uint8_t extension:1;
uint8_t padding:1;
uint8_t version:2;
/* byte 1 */
uint8_t payloadType:7;
uint8_t marker:1;
/* bytes 2,3 */
uint16_t seq;
/* bytes 4-7 */
uint32_t timestamp;
/* bytes 8-11 */
uint32_t ssrc;
};
struct RtpPacket
{
char header[4];
struct RtpHeader rtpHeader;
uint8_t payload[0];
};
void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
uint16_t seq, uint32_t timestamp, uint32_t ssrc);
int rtpSendPacket(int socket, uint8_t rtpChannel, struct RtpPacket* rtpPacket, uint32_t dataSize);
#endif //_RTP_H_
tcp_rtp.c
#include <sys/types.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include "tcp_rtp.h"
void rtpHeaderInit(struct RtpPacket* rtpPacket, uint8_t csrcLen, uint8_t extension,
uint8_t padding, uint8_t version, uint8_t payloadType, uint8_t marker,
uint16_t seq, uint32_t timestamp, uint32_t ssrc)
{
rtpPacket->rtpHeader.csrcLen = csrcLen;
rtpPacket->rtpHeader.extension = extension;
rtpPacket->rtpHeader.padding = padding;
rtpPacket->rtpHeader.version = version;
rtpPacket->rtpHeader.payloadType = payloadType;
rtpPacket->rtpHeader.marker = marker;
rtpPacket->rtpHeader.seq = seq;
rtpPacket->rtpHeader.timestamp = timestamp;
rtpPacket->rtpHeader.ssrc = ssrc;
}
int rtpSendPacket(int socket, uint8_t rtpChannel, struct RtpPacket* rtpPacket, uint32_t dataSize)
{
int ret;
rtpPacket->header[0] = '$';
rtpPacket->header[1] = rtpChannel;
rtpPacket->header[2] = ((dataSize+RTP_HEADER_SIZE) & 0xFF00 ) >> 8;
rtpPacket->header[3] = (dataSize+RTP_HEADER_SIZE) & 0xFF;
rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);
ret = send(socket, (void*)rtpPacket, dataSize+RTP_HEADER_SIZE+4, 0);
rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);
return ret;
}
将rtsp_server.c
、tcp_rtp.h
、tcp_rtp.c
保存下来
编译运行,程序默认打开test.h264
,如果你没有视频源的话,可以从RtspServer的example目录下获取
# gcc rtsp_server.c tcp_rtp.c
# ./a.out
运行后得到一个url
rtsp://127.0.0.1:8554
如何启动RTP OVER TCP?
需要设置vlc的模式
打开工具
>>首选项
>>输入/编解码器
>>live555 流传输
>>RTP over RTSP(TCP)
然后选择RTP over RTSP(TCP)
点击保存
输入url
运行效果
文章浏览阅读3.4k次,点赞8次,收藏42次。一、什么是内部类?or 内部类的概念内部类是定义在另一个类中的类;下面类TestB是类TestA的内部类。即内部类对象引用了实例化该内部对象的外围类对象。public class TestA{ class TestB {}}二、 为什么需要内部类?or 内部类有什么作用?1、 内部类方法可以访问该类定义所在的作用域中的数据,包括私有数据。2、内部类可以对同一个包中的其他类隐藏起来。3、 当想要定义一个回调函数且不想编写大量代码时,使用匿名内部类比较便捷。三、 内部类的分类成员内部_成员内部类和局部内部类的区别
文章浏览阅读118次。分布式系统要求拆分分布式思想的实质搭配要求分布式系统要求按照某些特定的规则将项目进行拆分。如果将一个项目的所有模板功能都写到一起,当某个模块出现问题时将直接导致整个服务器出现问题。拆分按照业务拆分为不同的服务器,有效的降低系统架构的耦合性在业务拆分的基础上可按照代码层级进行拆分(view、controller、service、pojo)分布式思想的实质分布式思想的实质是为了系统的..._分布式系统运维工具
文章浏览阅读174次。1.数据源准备2.数据处理step1:数据表处理应用函数:①VLOOKUP函数; ② CONCATENATE函数终表:step2:数据透视表统计分析(1) 透视表汇总不同渠道用户数, 金额(2)透视表汇总不同日期购买用户数,金额(3)透视表汇总不同用户购买订单数,金额step3:讲第二步结果可视化, 比如, 柱形图(1)不同渠道用户数, 金额(2)不同日期..._exce l趋势分析数据量
文章浏览阅读3.3k次。堡垒机可以为企业实现服务器、网络设备、数据库、安全设备等的集中管控和安全可靠运行,帮助IT运维人员提高工作效率。通俗来说,就是用来控制哪些人可以登录哪些资产(事先防范和事中控制),以及录像记录登录资产后做了什么事情(事后溯源)。由于堡垒机内部保存着企业所有的设备资产和权限关系,是企业内部信息安全的重要一环。但目前出现的以下问题产生了很大安全隐患:密码设置过于简单,容易被暴力破解;为方便记忆,设置统一的密码,一旦单点被破,极易引发全面危机。在单一的静态密码验证机制下,登录密码是堡垒机安全的唯一_horizon宁盾双因素配置
文章浏览阅读7.7k次,点赞4次,收藏16次。Chrome作为一款挺不错的浏览器,其有着诸多的优良特性,并且支持跨平台。其支持(Windows、Linux、Mac OS X、BSD、Android),在绝大多数情况下,其的安装都很简单,但有时会由于网络原因,无法安装,所以在这里总结下Chrome的安装。Windows下的安装:在线安装:离线安装:Linux下的安装:在线安装:离线安装:..._chrome linux debian离线安装依赖
文章浏览阅读153次。中国发达城市榜单每天都在刷新,但无非是北上广轮流坐庄。北京拥有最顶尖的文化资源,上海是“摩登”的国际化大都市,广州是活力四射的千年商都。GDP和发展潜力是衡量城市的数字指...
文章浏览阅读3.3k次。前言spark在java使用比较少,多是scala的用法,我这里介绍一下我在项目中使用的代码配置详细算法的使用请点击我主页列表查看版本jar版本说明spark3.0.1scala2.12这个版本注意和spark版本对应,只是为了引jar包springboot版本2.3.2.RELEASEmaven<!-- spark --> <dependency> <gro_使用java调用spark注册进去的程序
文章浏览阅读4.8k次。汽车零部件开发工具巨头V公司全套bootloader中UDS协议栈源代码,自己完成底层外设驱动开发后,集成即可使用,代码精简高效,大厂出品有量产保证。:139800617636213023darcy169_uds协议栈 源代码
文章浏览阅读4.6k次,点赞20次,收藏148次。AUTOSAR基础篇之OS(下)前言首先,请问大家几个小小的问题,你清楚:你知道多核OS在什么场景下使用吗?多核系统OS又是如何协同启动或者关闭的呢?AUTOSAR OS存在哪些功能安全等方面的要求呢?多核OS之间的启动关闭与单核相比又存在哪些异同呢?。。。。。。今天,我们来一起探索并回答这些问题。为了便于大家理解,以下是本文的主题大纲:[外链图片转存失败,源站可能有防盗链机制,建议将图片保存下来直接上传(img-JCXrdI0k-1636287756923)(https://gite_autosar 定义了 5 种多核支持类型
文章浏览阅读2.2k次,点赞6次,收藏14次。原因:自己写的头文件没有被加入到方案的包含目录中去,无法被检索到,也就无法打开。将自己写的头文件都放入header files。然后在VS界面上,右键方案名,点击属性。将自己头文件夹的目录添加进去。_vs2013打不开自己定义的头文件
文章浏览阅读3.3w次,点赞80次,收藏342次。此时,可以将系统中所有用户的 Session 数据全部保存到 Redis 中,用户在提交新的请求后,系统先从Redis 中查找相应的Session 数据,如果存在,则再进行相关操作,否则跳转到登录页面。此时,可以将系统中所有用户的 Session 数据全部保存到 Redis 中,用户在提交新的请求后,系统先从Redis 中查找相应的Session 数据,如果存在,则再进行相关操作,否则跳转到登录页面。当数据量很大时,count 的数量的指定可能会不起作用,Redis 会自动调整每次的遍历数目。_redis命令
文章浏览阅读449次,点赞3次,收藏3次。URP的设计目标是在保持高性能的同时,提供更多的渲染功能和自定义选项。与普通项目相比,会多出Presets文件夹,里面包含着一些设置,包括本色,声音,法线,贴图等设置。全局只有主光源和附加光源,主光源只支持平行光,附加光源数量有限制,主光源和附加光源在一次Pass中可以一起着色。URP:全局只有主光源和附加光源,主光源只支持平行光,附加光源数量有限制,一次Pass可以计算多个光源。可编程渲染管线:渲染策略是可以供程序员定制的,可以定制的有:光照计算和光源,深度测试,摄像机光照烘焙,后期处理策略等等。_urp渲染管线