从零开始写一个RTSP服务器系列
从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器
从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器
从零开始写一个RTSP服务器（九）一个RTP OVER RTSP/TCP的RTSP服务器
从零开始写一个RTSP服务器（六）一个传输AAC的RTSP服务器
文章目录
本文主要是结合
本文主要是结合
这几篇文章的内容总结，然后写出完成一个传输AAC的RTSP服务器，所有的知识点都在这几篇文章中，在看此篇文章前建议先认真看一看前面那几篇文章
本文不会讲解新知识点，如果你仔细看前几篇文章，相信你不需要看本文就能够完成一个传输AAC的RTSP服务器
下面主要是介绍我提供的示例代码的运行流程和其中的一些细节
一、建立套接字
一开始进入main函数后，就监听服务器tcp套接字，绑定端口号，然后开始监听
然后再分别建立用于RTP和RTCP的udp套接字，绑定好端口
然后进入循环中开始服务
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
main()
{
/* 创建服务器tcp套接字，绑定端口，监听 */
serverSockfd = createTcpSocket();
bindSocketAddr(serverSockfd, "0.0.0.0", SERVER_PORT);
listen(serverSockfd, 10);
/* 建立用于RTP和RTCP的udp套接字，绑定好端口 */
serverRtpSockfd = createUdpSocket();
serverRtcpSockfd = createUdpSocket();
bindSocketAddr(serverRtpSockfd, "0.0.0.0", SERVER_RTP_PORT);
bindSocketAddr(serverRtcpSockfd, "0.0.0.0", SERVER_RTCP_PORT);
while(1)
{
...
}
}
二、接收客户端连接
在while循环中接收客户端，然后调用doClient服务
main()
{
...
while(1)
{
clientSockfd = acceptClient(serverSockfd, clientIp, &clientPort);
doClient(clientSockfd, clientIp, clientPort, serverRtpSockfd, serverRtcpSockfd);
}
}
上面其实就是一个TCP服务器的基本步骤，没有什么特别的
下面来看一看doClient函数
三、解析命令
doClient就是一个while循环（这是一个同时只能服务一个客户的服务器），不断地接收命令解析命令，然后调用相应地操作
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
doClient()
{
while(1)
{
recv(clientSockfd, rBuf, BUF_MAX_SIZE, 0);
...
sscanf(line, "%s %s %s\r\n", method, url, version);
...
sscanf(line, "CSeq: %d\r\n", &cseq)
...
}
}
四、处理请求
在解析完客户端命令后，会调用相应的请求，处理完之后讲接收打印到sBuf中，然后发送给客户端
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
doClient()
{
while(1)
{
...
/* 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, clientRtpPort);
else if(!strcmp(method, "PLAY"))
handleCmd_PLAY(sBuf, cseq);
/* 放回结果 */
send(clientSockfd, sBuf, strlen(sBuf), 0);
}
}
下面来看看各个请求的行动
4.1 OPTIONS
返回可用方法
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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;
}
4.2 DESCRIBE
返回sdp文件信息，注意这个示例的sdp文件和从零开始写一个RTSP服务器（四）一个传输H.264的RTSP服务器中的sdp文件是不一样的，这是很重要的
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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 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;
}
4.3 SETUP
SETUP过程发送服务端RTP端口和RTCP端口
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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;
}
4.4 PLAY
PLAY操作回复后，会开始发送RTP包
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
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;
}
五、AAC RTP打包发送
先读取ADTS头，得到一帧的大小，然后再读取AAC Data，再通过RTP打包传输
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
doClient()
{
while(1)
{
...
send(clientSockfd, sBuf, strlen(sBuf), 0);
if(!strcmp(method, "PLAY"))
{
while(1)
{
/* 读取ADTS头部 */
read(fd, frame, 7);
/* 解析头部 */
parseAdtsHeader(frame, &adtsHeader);
/* 读取一帧 */
read(fd, frame, adtsHeader.aacFrameLength-7);
/* RTP打包发送 */
rtpSendAACFrame(localRtpSockfd, clientIP, clientRtpPort,
rtpPacket, frame, adtsHeader.aacFrameLength-7);
}
}
}
}
看一看AAC的RTP打包发送过程
先填充RTP载荷前4个字节，然后发送RTP包
发送后序列号增加，时间戳增加
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
static int rtpSendAACFrame(int socket, const char* ip, int16_t port,
struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize)
{
/* 填充前4个字节 */
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
/* 发送RTP包 */
memcpy(rtpPacket->payload+4, frame, frameSize);
rtpSendPacket(socket, ip, port, rtpPacket, frameSize+4);
/* 增加序列号和时间戳 */
rtpPacket->rtpHeader.seq++;
rtpPacket->rtpHeader.timestamp += 1025;
}
六、源码
总共由三个文件aac_rtsp_server.c、rtp.h、rtp.c
aac_rtsp_server.c
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#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 "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 "test.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'，说明一个ADTS帧的开始
unsigned int id; //1 bit MPEG 标示符， 0 for MPEG-4，1 for MPEG-2
unsigned int layer; //2 bit 总是'00'
unsigned int protectionAbsent; //1 bit 1表示没有crc，0表示有crc
unsigned int profile; //1 bit 表示使用哪个级别的AAC
unsigned int samplingFreqIndex; //4 bit 表示使用的采样频率
unsigned int privateBit; //1 bit
unsigned int channelCfg; //3 bit 表示声道数
unsigned int originalCopy; //1 bit
unsigned int home; //1 bit
/*下面的为改变的参数即每一帧都不同*/
unsigned int copyrightIdentificationBit; //1 bit
unsigned int copyrightIdentificationStart; //1 bit
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 0x7FF 说明是码率可变的码流
/* 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)
*/
unsigned int 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 = ((unsigned int) in[1] & 0x08) >> 3;
res->layer = ((unsigned int) in[1] & 0x06) >> 1;
res->protectionAbsent = (unsigned int) in[1] & 0x01;
res->profile = ((unsigned int) in[2] & 0xc0) >> 6;
res->samplingFreqIndex = ((unsigned int) in[2] & 0x3c) >> 2;
res->privateBit = ((unsigned int) in[2] & 0x02) >> 1;
res->channelCfg = ((((unsigned int) in[2] & 0x01) << 2) | (((unsigned int) in[3] & 0xc0) >> 6));
res->originalCopy = ((unsigned int) in[3] & 0x20) >> 5;
res->home = ((unsigned int) in[3] & 0x10) >> 4;
res->copyrightIdentificationBit = ((unsigned int) in[3] & 0x08) >> 3;
res->copyrightIdentificationStart = (unsigned int) 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 = ((unsigned int) 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)
{
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 = rtpSendPacket(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 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=60\r\n\r\n",
cseq);
return 0;
}
static void doClient(int clientSockfd, const char* clientIP, int clientPort,
int serverRtpSockfd, int serverRtcppSockfd)
{
char method[40];
char url[100];
char version[40];
int cseq;
int clientRtpPort, clientRtcpPort;
char *bufPtr;
char* rBuf = malloc(BUF_MAX_SIZE);
char* sBuf = malloc(BUF_MAX_SIZE);
char line[400];
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client_port */
if(!strcmp(method, "SETUP"))
{
while(1)
{
bufPtr = getLineFromBuf(bufPtr, line);
if(!strncmp(line, "Transport:", strlen("Transport:")))
{
sscanf(line, "Transport: RTP/AVP;unicast;client_port=%d-%d\r\n",
&clientRtpPort, &clientRtcpPort);
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, clientRtpPort))
{
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);
if(!strcmp(method, "PLAY"))
{
struct AdtsHeader adtsHeader;
struct RtpPacket* rtpPacket;
uint8_t* frame;
int ret;
int fd = open(AAC_FILE_NAME, O_RDONLY);
if(fd < 0)
{
printf("failed to open %s\n", AAC_FILE_NAME);
goto out;
}
frame = (uint8_t*)malloc(5000);
rtpPacket = malloc(5000);
rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_AAC, 1, 0, 0, 0x32411);
while(1)
{
ret = read(fd, frame, 7);
if(ret <= 0)
{
break;
}
if(parseAdtsHeader(frame, &adtsHeader) < 0)
{
printf("parse err\n");
break;
}
ret = read(fd, frame, adtsHeader.aacFrameLength-7);
if(ret < 0)
{
printf("read err\n");
break;
}
rtpSendAACFrame(serverRtpSockfd, clientIP, clientRtpPort,
rtpPacket, frame, adtsHeader.aacFrameLength-7);
usleep(23000);
}
free(frame);
free(rtpPacket);
}
}
out:
close(clientSockfd);
free(rBuf);
free(sBuf);
}
int main(int argc, char* argv[])
{
int serverSockfd;
int serverRtpSockfd, serverRtcpSockfd;
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;
}
serverRtpSockfd = createUdpSocket();
serverRtcpSockfd = createUdpSocket();
if(serverRtpSockfd < 0 || serverRtcpSockfd < 0)
{
printf("failed to create udp socket\n");
return -1;
}
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");
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, serverRtpSockfd, serverRtcpSockfd);
}
return 0;
}
rtp.h
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#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
{
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, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize);
#endif //_RTP_H_
rtp.c
/*
* 作者：_JT_
* 博客：https://blog.csdn.net/weixin_42462202
*/
#include <sys/types.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include "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, const char* ip, int16_t port, struct RtpPacket* rtpPacket, uint32_t dataSize)
{
struct sockaddr_in addr;
int ret;
addr.sin_family = AF_INET;
addr.sin_port = htons(port);
addr.sin_addr.s_addr = inet_addr(ip);
rtpPacket->rtpHeader.seq = htons(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = htonl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = htonl(rtpPacket->rtpHeader.ssrc);
ret = sendto(socket, (void*)rtpPacket, dataSize+RTP_HEADER_SIZE, 0,
(struct sockaddr*)&addr, sizeof(addr));
rtpPacket->rtpHeader.seq = ntohs(rtpPacket->rtpHeader.seq);
rtpPacket->rtpHeader.timestamp = ntohl(rtpPacket->rtpHeader.timestamp);
rtpPacket->rtpHeader.ssrc = ntohl(rtpPacket->rtpHeader.ssrc);
return ret;
}
七、测试
将aac_rtsp_server.c、rtp.h、rtp.c这三个文件保存下来
编译运行，默认会打开test.aac的音频文件，如果你没有音频源的话，可以从RtspServer的example目录下获取
# gcc aac_rtsp_server.c rtp.c
# ./a.out
运行之后会打印一个url
rtsp://127.0.0.1:8554
再vlc打开即可听到音频
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