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2w次,点赞7次,收藏51次。四个步骤1.创建C++ Win32项目动态库dll 2.在Win32项目动态库中添加 外部依赖项 lib头文件和lib库3.导出C接口4.c#调用c++动态库开始你的表演...①创建一个空白的解决方案,在解决方案中添加 Visual C++ , Win32 项目空白解决方案的创建:添加Visual C++ , Win32 项目这......_c#调用lib
文章浏览阅读4.6k次。苹方字体是苹果系统上的黑体,挺好看的。注重颜值的网站都会使用,例如知乎:font-family: -apple-system, BlinkMacSystemFont, Helvetica Neue, PingFang SC, Microsoft YaHei, Source Han Sans SC, Noto Sans CJK SC, W..._ubuntu pingfang
文章浏览阅读159次。表单表单概述表单标签表单域按钮控件demo表单标签表单标签基本语法结构<form action="处理数据程序的url地址“ method=”get|post“ name="表单名称”></form><!--action,当提交表单时,向何处发送表单中的数据,地址可以是相对地址也可以是绝对地址--><!--method将表单中的数据传送给服务器处理,get方式直接显示在url地址中,数据可以被缓存,且长度有限制;而post方式数据隐藏传输,_html表单的处理程序有那些
文章浏览阅读1.2k次。使用说明:开启Google的登陆二步验证(即Google Authenticator服务)后用户登陆时需要输入额外由手机客户端生成的一次性密码。实现Google Authenticator功能需要服务器端和客户端的支持。服务器端负责密钥的生成、验证一次性密码是否正确。客户端记录密钥后生成一次性密码。下载谷歌验证类库文件放到项目合适位置(我这边放在项目Vender下面)https://github.com/PHPGangsta/GoogleAuthenticatorPHP代码示例://引入谷_php otp 验证器
文章浏览阅读4.3k次,点赞5次,收藏11次。matplotlib.plot画图横坐标混乱及间隔处理_matplotlib更改横轴间距
文章浏览阅读2.2k次。①Storage driver 处理各镜像层及容器层的处理细节,实现了多层数据的堆叠,为用户 提供了多层数据合并后的统一视图②所有 Storage driver 都使用可堆叠图像层和写时复制(CoW)策略③docker info 命令可查看当系统上的 storage driver主要用于测试目的,不建议用于生成环境。_docker 保存容器
文章浏览阅读834次,点赞27次,收藏13次。网络拓扑结构是指计算机网络中各组件(如计算机、服务器、打印机、路由器、交换机等设备)及其连接线路在物理布局或逻辑构型上的排列形式。这种布局不仅描述了设备间的实际物理连接方式,也决定了数据在网络中流动的路径和方式。不同的网络拓扑结构影响着网络的性能、可靠性、可扩展性及管理维护的难易程度。_网络拓扑csdn
文章浏览阅读1.8k次,点赞5次,收藏8次。IOS系统Date的坑要创建一个指定时间的new Date对象时,通常的做法是:new Date("2020-09-21 11:11:00")这行代码在 PC 端和安卓端都是正常的,而在 iOS 端则会提示 Invalid Date 无效日期。在IOS年月日中间的横岗许换成斜杠,也就是new Date("2020/09/21 11:11:00")通常为了兼容IOS的这个坑,需要做一些额外的特殊处理,笔者在开发的时候经常会忘了兼容IOS系统。所以就想试着重写Date函数,一劳永逸,避免每次ne_date.prototype 将所有 ios
文章浏览阅读5.3k次。方法一:用PLSQL Developer工具。 1 在PLSQL Developer的sql window里输入select * from test for update; 2 按F8执行 3 打开锁, 再按一下加号. 鼠标点到第一列的列头,使全列成选中状态,然后粘贴,最后commit提交即可。(前提..._excel导入pl/sql
文章浏览阅读83次。Git常用命令速查手册1、初始化仓库git init2、将文件添加到仓库git add 文件名 # 将工作区的某个文件添加到暂存区 git add -u # 添加所有被tracked文件中被修改或删除的文件信息到暂存区,不处理untracked的文件git add -A # 添加所有被tracked文件中被修改或删除的文件信息到暂存区,包括untracked的文件...
文章浏览阅读202次。分享119个ASP.NET源码总有一个是你想要的_千博二手车源码v2023 build 1120
文章浏览阅读1.8k次。版权声明:转载请注明出处 http://blog.csdn.net/irean_lau。目录(?)[+]1、缺省构造函数。2、缺省拷贝构造函数。3、 缺省析构函数。4、缺省赋值运算符。5、缺省取址运算符。6、 缺省取址运算符 const。[cpp] view plain copy_空类默认产生哪些类成员函数