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4.4k次,点赞2次,收藏16次。写这篇文章的由来是因为后边要用这个工具,但是由于某些原因有部分小伙伴和童鞋们可能不会安装此工具,为了方便小伙伴们和童鞋们的后续学习和不打击他们的积极性,因为80%的人都是死在工具的安装这第一道门槛上,这门槛说高也不高说低也不是太低。所以就抽时间水了这一篇文章。_eclipse安装教程
文章浏览阅读4.1w次,点赞12次,收藏193次。小编为大家收集了11个web前端开发,大企业实战项目案例+5W行源码!拿走玩去吧!1)小米官网项目描述:首先选择小米官网为第一个实战案例,是因为刚开始入门,有个参考点,另外站点比较偏向目前的卡片式设计,实现常见效果。目的为学者练习编写小米官网,熟悉div+css布局。学习资料的话可以加下web前端开发学习裙:600加上610再加上151自己去群里下载下。项目技术:HTML+CSS+Div布局2)迅雷官网项目描述:此站点特效较多,所以通过练习编写次站点,学生可以更多练习CSS3的新特性过渡与动画的实_前端项目实战案例
文章浏览阅读73次。素数,不同的质数,各种各样的问题总是遇到的素数。以下我们来说一下求素数的一种比較有效的算法。就是筛法。由于这个要求得1-n区间的素数仅仅须要O(nloglogn)的时间复杂度。以下来说一下它的思路。思路:如今又1-n的数字。素数嘛就是除了1和本身之外没有其它的约数。所以有约数的都不是素数。我们从2開始往后遍历,是2的倍数的都不是素数。所以我们把他们划掉然后如...
文章浏览阅读532次,点赞9次,收藏14次。探索Keras DCGAN:深度学习中的创新图像生成项目地址:https://gitcode.com/jacobgil/keras-dcgan在数据驱动的时代,图像生成模型已经成为人工智能的一个重要领域。其中,Keras DCGAN 是一个基于 Keras 的实现,用于构建和训练 Deep Convolutional Generative Adversarial Networks(深度卷积生...
文章浏览阅读116次。今天在搭建springcloud项目时,发现如上错误,顺便整理一下这个异常:1. mapper.xml的命名空间(namespace)是否跟mapper的接口路径一致<mapper namespace="com.baicun.springcloudprovider.mapper.SysUserMapper">2.mapper.xml接口名是否和mapper.java接..._spring-could org.apache.ibatis.binding.bindingexception: invalid bound state
文章浏览阅读1.1k次。四种高效数据库设计思想——提高查询效率:设计数据库表结构时,我们首先要按照数据库的三大范式进行建立数据。1. 1NF每列不可拆分2. 2NF确保每个表只做一件事情3. 3NF满足2NF,消除表中的依赖传递。三大范式的出现是在上世纪70年代,由于内存资源比较昂贵,所以严格按照三大范式进行数据库设计。而如今内存变得越来越廉价,在考虑效率和内存的基础上我们可以做出最优选择以达到最高效率。_数据库为什么能提高效率
文章浏览阅读1.6k次。应用程序在启动和运行的时候往往需要读取一些配置信息,配置基本上伴随着应用程序的整个生命周期,比如:数 据库连接参数、启动参数等。配置主要有以下几个特点:配置是独立于程序的只读变量配置对于程序是只读的,程序通过读取配置来改变自己的行为,但是程序不应该去改变配置配置伴随应用的整个生命周期配置贯穿于应用的整个生命周期,应用在启动时通过读取配置来初始化,在运行时根据配置调整行为。比如:启动时需要读取服务的端口号、系统在运行过程中需要读取定时策略执行定时任务等。配置可以有多种加载方式常见的有程序内部_基于配置是什么意思
文章浏览阅读170次。Glib库实现了一个非常重要的基础类--GObject,这个类中封装了许多我们在定义和实现类时经常用到的机制: 引用计数式的内存管理 对象的构造与析构 通用的属性(Property)机制 Signal的简单使用方式 很多使用GObject..._
文章浏览阅读6.3k次,点赞2次,收藏9次。在 golang 中若写定时脚本,有两种实现。一、基于原生语法组装func DocSyncTaskCronJob() { ticker := time.NewTicker(time.Minute * 5) // 每分钟执行一次 for range ticker.C { ProcTask() }}func ProcTask() { log.Println("hello world")}二、基于 github 中封装的 cron 库实现package taskimport (_golang 定时任务
文章浏览阅读2.1k次。 来源:http://blog.csdn.net/clever101/archive/2008/10/18/3096049.aspx 声明:本文章是我整合网上的资料而成的,其中的大部分文字不是我所为的,我所起的作用只是归纳整理并添加我的一些看法。非常感谢引用到的文字的作者的辛勤劳动,所参考的文献在文章最后我已一一列出。 对关注性能的程序开发人员而言,一个好的计时部件既是益友,也_vc 通过线程和 sleep 获取精准时间
文章浏览阅读58次。公司突然说要进行wap开发了,以前从没了解过,但我却异常的兴奋,因为可以学习新东西了,呵呵,我们大家一起努力吧。首先说说环境的搭建。可以把.wml的文件看做是另一种的html进行信息的展示,但并不是所有的浏览器都支持,好用的有Opera,还有WinWap。编写wml文件语法比较严格,不好的是我还没有找到好的提示工具,就先用纯文本吧。我找到了一个很好的学习网站:http://w3sc..._winwap学习
文章浏览阅读504次。考研成绩出来后,第一件事是干什么?当然不只是高兴,而是马上给心仪的导师发邮件,先露个“名字熟”。不要以为初试考了高分或者过线了,一切都稳妥了,一时得意忘形,居然没联系导师,等想起时,导师已经属于他人了。对于一些大佬,热门导师一定要趁早发邮件咨询,一是表示尊重;二是这类老师可能已经没有统招名额,所以越早知道,越有利于下一步计划。但是,在给导师发邮件中,要注意以下4点,不求一步成功,但求先留下个好印象..._跨考计算机怎么给导师发邮件