模仿b站北小菜的教学,学习了基于udp或tcp的rtp传输acc或者h264的rtsp服务器。
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>//定义了uint8_t是什么 原来的c整型有缺点 uint在循环中可以不停止
#include <string.h>
#include <time.h>
#include <sys/types.h>//系统数据类型
#include <sys/stat.h>//系统属性
#include <fcntl.h>//文件的打开、数据写入、数据读取、关闭文件的操作
#include <WinSock2.h>
#include <WS2tcpip.h>
#include <windows.h>
#include "rtp.h"
#define SERVER_PORT 8554
#define SERVER_RTP_PORT 55532
#define SERVER_RTCP_PORT 55533
#define BUF_MAX_SIZE (1024*1024)
#define AAC_FILE_NAME "../data/test-long.aac"
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;
}
struct AdtsHeader {
unsigned int syncword; //12 bit 同步字 '1111 1111 1111',一个ADTS帧的开始
uint8_t id; //1 bit 0代表MPEG-4, 1代表MPEG-2。
uint8_t layer; //2 bit 必须为0
uint8_t protectionAbsent; //1 bit 1代表没有CRC,0代表有CRC
uint8_t profile; //1 bit AAC级别(MPEG-2 AAC中定义了3种profile,MPEG-4 AAC中定义了6种profile)
uint8_t samplingFreqIndex; //4 bit 采样率
uint8_t privateBit; //1bit 编码时设置为0,解码时忽略
uint8_t channelCfg; //3 bit 声道数量
uint8_t originalCopy; //1bit 编码时设置为0,解码时忽略
uint8_t home; //1 bit 编码时设置为0,解码时忽略
uint8_t copyrightIdentificationBit; //1 bit 编码时设置为0,解码时忽略
uint8_t copyrightIdentificationStart; //1 bit 编码时设置为0,解码时忽略
unsigned int aacFrameLength; //13 bit 一个ADTS帧的长度包括ADTS头和AAC原始流
unsigned int adtsBufferFullness; //11 bit 缓冲区充满度,0x7FF说明是码率可变的码流,不需要此字段。CBR可能需要此字段,不同编码器使用情况不同。这个在使用音频编码的时候需要注意。
/* number_of_raw_data_blocks_in_frame
* 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧
* 所以说number_of_raw_data_blocks_in_frame == 0
* 表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据)
*/
uint8_t numberOfRawDataBlockInFrame; //2 bit
};
static int parseAdtsHeader(uint8_t* in, struct AdtsHeader* res) {
static int frame_number = 0;
memset(res, 0, sizeof(*res));
if ((in[0] == 0xFF) && ((in[1] & 0xF0) == 0xF0))
{
res->id = ((uint8_t)in[1] & 0x08) >> 3;//第二个字节与0x08与运算之后,获得第13位bit对应的值
res->layer = ((uint8_t)in[1] & 0x06) >> 1;//第二个字节与0x06与运算之后,右移1位,获得第14,15位两个bit对应的值
res->protectionAbsent = (uint8_t)in[1] & 0x01;
res->profile = ((uint8_t)in[2] & 0xc0) >> 6;
res->samplingFreqIndex = ((uint8_t)in[2] & 0x3c) >> 2;
res->privateBit = ((uint8_t)in[2] & 0x02) >> 1;
res->channelCfg = ((((uint8_t)in[2] & 0x01) << 2) | (((unsigned int)in[3] & 0xc0) >> 6));
res->originalCopy = ((uint8_t)in[3] & 0x20) >> 5;
res->home = ((uint8_t)in[3] & 0x10) >> 4;
res->copyrightIdentificationBit = ((uint8_t)in[3] & 0x08) >> 3;
res->copyrightIdentificationStart = (uint8_t)in[3] & 0x04 >> 2;
res->aacFrameLength = (((((unsigned int)in[3]) & 0x03) << 11) |
(((unsigned int)in[4] & 0xFF) << 3) |
((unsigned int)in[5] & 0xE0) >> 5);
res->adtsBufferFullness = (((unsigned int)in[5] & 0x1f) << 6 |
((unsigned int)in[6] & 0xfc) >> 2);
res->numberOfRawDataBlockInFrame = ((uint8_t)in[6] & 0x03);
return 0;
}
else
{
printf("failed to parse adts header\n");
return -1;
}
}
static int rtpSendAACFrame(int socket, const char* ip, int16_t port,
struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize) {
//打包文档:https://blog.csdn.net/yangguoyu8023/article/details/106517251/
int ret;
rtpPacket->payload[0] = 0x00;
rtpPacket->payload[1] = 0x10;
rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; //高8位
rtpPacket->payload[3] = (frameSize & 0x1F) << 3; //低5位
memcpy(rtpPacket->payload + 4, frame, frameSize);
ret = rtpSendPacketOverUdp(socket, ip, port, rtpPacket, frameSize + 4);
if (ret < 0)
{
printf("failed to send rtp packet\n");
return -1;
}
rtpPacket->rtpHeader.seq++;
/*
* 如果采样频率是44100
* 一般AAC每个1024个采样为一帧
* 所以一秒就有 44100 / 1024 = 43帧
* 时间增量就是 44100 / 43 = 1025
* 一帧的时间为 1 / 43 = 23ms
*/
rtpPacket->rtpHeader.timestamp += 1025;
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 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=audio 0 RTP/AVP 97\r\n"
"a=rtpmap:97 mpeg4-generic/44100/2\r\n"
"a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1210;\r\n"
//"a=fmtp:97 SizeLength=13;\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, int clientRtpPort) {
sprintf(result, "RTSP/1.0 200 OK\r\n"
"CSeq: %d\r\n"
"Transport: RTP/AVP;unicast;client_port=%d-%d;server_port=%d-%d\r\n"
"Session: 66334873\r\n"
"\r\n",
cseq,
clientRtpPort,
clientRtpPort + 1,
SERVER_RTP_PORT,
SERVER_RTCP_PORT
);
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=10\r\n\r\n",
cseq);
return 0;
}
static void doClient(int clientSockfd, const char* clientIP, int clientPort) {
int serverRtpSockfd = -1, serverRtcpSockfd = -1;
char method[40];
char url[100];
char version[40];
int CSeq;
int clientRtpPort, clientRtcpPort;
char* rBuf = (char*)malloc(BUF_MAX_SIZE);
char* sBuf = (char*)malloc(BUF_MAX_SIZE);
while (true) {
int recvLen;
recvLen = recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
if (recvLen <= 0) {
break;
}
rBuf[recvLen] = '\0';
printf("%s rBuf = %s \n", __FUNCTION__, rBuf);
const char* sep = "\n";
char* line = strtok(rBuf, sep);
while (line) {
if (strstr(line, "OPTIONS") ||
strstr(line, "DESCRIBE") ||
strstr(line, "SETUP") ||
strstr(line, "PLAY")) {
if (sscanf(line, "%s %s %s\r\n", method, url, version) != 3) {
// error
}
}
else if (strstr(line, "CSeq")) {
if (sscanf(line, "CSeq: %d\r\n", &CSeq) != 1) {
// error
}
}
else if (!strncmp(line, "Transport:", strlen("Transport:"))) {
// Transport: RTP/AVP/UDP;unicast;client_port=13358-13359
// Transport: RTP/AVP;unicast;client_port=13358-13359
if (sscanf(line, "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d\r\n",
&clientRtpPort, &clientRtcpPort) != 2) {
// error
printf("parse Transport error \n");
}
}
line = strtok(NULL, sep);
}
if (!strcmp(method, "OPTIONS")) {
if (handleCmd_OPTIONS(sBuf, CSeq))
{
printf("failed to handle options\n");
break;
}
}
else if (!strcmp(method, "DESCRIBE")) {
if (handleCmd_DESCRIBE(sBuf, CSeq, url))
{
printf("failed to handle describe\n");
break;
}
}
else if (!strcmp(method, "SETUP")) {
if (handleCmd_SETUP(sBuf, CSeq, clientRtpPort))
{
printf("failed to handle setup\n");
break;
}
serverRtpSockfd = createUdpSocket();
serverRtcpSockfd = createUdpSocket();
if (serverRtpSockfd < 0 || serverRtcpSockfd < 0)
{
printf("failed to create udp socket\n");
break;
}
if (bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT) < 0 ||
bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT) < 0)
{
printf("failed to bind addr\n");
break;
}
}
else if (!strcmp(method, "PLAY")) {
if (handleCmd_PLAY(sBuf, CSeq))
{
printf("failed to handle play\n");
break;
}
}
else {
printf("未定义的method = %s \n", method);
break;
}
printf("%s sBuf = %s \n", __FUNCTION__, sBuf);
send(clientSockfd, sBuf, strlen(sBuf), 0);
//开始播放,发送RTP包
if (!strcmp(method, "PLAY")) {
struct AdtsHeader adtsHeader;
struct RtpPacket* rtpPacket;
uint8_t* frame;
int ret;
FILE* fp = fopen(AAC_FILE_NAME, "rb");
if (!fp) {
printf("读取 %s 失败\n", AAC_FILE_NAME);
break;
}
frame = (uint8_t*)malloc(5000);
rtpPacket = (struct RtpPacket*)malloc(5000);
rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_AAC, 1, 0, 0, 0x32411);
while (true)
{
ret = fread(frame, 1, 7, fp);
if (ret <= 0)
{
printf("fread err\n");
break;
}
printf("fread ret=%d \n",ret);
if (parseAdtsHeader(frame, &adtsHeader) < 0)
{
printf("parseAdtsHeader err\n");
break;
}
ret = fread(frame, 1, adtsHeader.aacFrameLength - 7, fp);
if (ret <= 0)
{
printf("fread err\n");
break;
}
rtpSendAACFrame(serverRtpSockfd, clientIP, clientRtpPort,
rtpPacket, frame, adtsHeader.aacFrameLength - 7);
Sleep(1);
//usleep(23223);//1000/43.06 * 1000
}
free(frame);
free(rtpPacket);
break;
}
memset(method, 0, sizeof(method) / sizeof(char));
memset(url, 0, sizeof(url) / sizeof(char));
CSeq = 0;
}
closesocket(clientSockfd);
if (serverRtpSockfd) {
closesocket(serverRtpSockfd);
}
if (serverRtcpSockfd > 0) {
closesocket(serverRtcpSockfd);
}
free(rBuf);
free(sBuf);
}
int main() {
// 启动windows socket start
WSADATA wsaData;
if (WSAStartup(MAKEWORD(2, 2), &wsaData) != 0)
{
printf("PC Server Socket Start Up Error \n");
return -1;
}
// 启动windows socket end
int rtspServerSockfd;
int ret;
rtspServerSockfd = createTcpSocket();
if (rtspServerSockfd < 0)
{
printf("failed to create tcp socket\n");
return -1;
}
ret = bindSocketAddr(rtspServerSockfd, "0.0.0.0", SERVER_PORT);
if (ret < 0)
{
printf("failed to bind addr\n");
return -1;
}
ret = listen(rtspServerSockfd, 10);
if (ret < 0)
{
printf("failed to listen\n");
return -1;
}
printf("%s rtsp://127.0.0.1:%d\n", __FILE__, SERVER_PORT);
while (1)
{
int clientSockfd;
char clientIp[40];
int clientPort;
clientSockfd = acceptClient(rtspServerSockfd, 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);
}
closesocket(rtspServerSockfd);
return 0;
}
用来记录,没有附上完全的代码,因为不会上传文件