diff --git a/code/aac_rtsp/Makefile b/code/aac_rtsp/Makefile new file mode 100644 index 0000000000000000000000000000000000000000..b7981f5fa9d979a7353e3967b6dd54ee588dc029 --- /dev/null +++ b/code/aac_rtsp/Makefile @@ -0,0 +1,4 @@ +main:main.c rtp.c rtp.h + gcc main.c rtp.c rtp.h -o main +clean: + rm -f main \ No newline at end of file diff --git a/code/aac_rtsp/README.md b/code/aac_rtsp/README.md new file mode 100644 index 0000000000000000000000000000000000000000..72eda2c815aa1c465deaf17ecdd9af6e0dce8b08 --- /dev/null +++ b/code/aac_rtsp/README.md @@ -0,0 +1,23 @@ +# Complie Only + +- Generate binary executable file +``` + make +``` + +- Clear binary executable file +``` + make clean +``` + +- Compile and execute +``` + ./build_and_run.sh +``` + +#### After executing an executable binary file and obtaining the RTSP stream address,ffplay can be used to play directly. +- The playback command is +``` + ffplay -i rtsp://192.168.50.236 +``` +- Modify IP address based on actual situation. \ No newline at end of file diff --git a/code/aac_rtsp/build_and_run.sh b/code/aac_rtsp/build_and_run.sh new file mode 100755 index 0000000000000000000000000000000000000000..27aee650035b61925a6898efa603c416a927bc1d --- /dev/null +++ b/code/aac_rtsp/build_and_run.sh @@ -0,0 +1,12 @@ +#!/bin/bash + +# run make command to compile create goal file +make + +# check if the target file has been generated +if [ -f main ]; then + #execute the generated main executable file + ./main +else + echo "fail to compile, can not generated main executable file." +fi \ No newline at end of file diff --git a/code/aac_rtsp/main.c b/code/aac_rtsp/main.c new file mode 100644 index 0000000000000000000000000000000000000000..c0f655e79069f4c7b6434f3ddb09360d17c2c929 --- /dev/null +++ b/code/aac_rtsp/main.c @@ -0,0 +1,517 @@ +// +// gcc main.c rtp.c rtp.h -o main +// ./main test.aac +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "rtp.h" + +#define SERVER_IP "192.168.50.236" +#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; +} + +// Use to describe AAC audio data +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 + + //将aac音频帧的数据复制到RTP包的有效负载(payload)部分的适当位置 + 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 (1) + { + 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, SERVER_IP, SERVER_RTP_PORT) < 0 || + bindSocketAddr(serverRtcpSockfd, SERVER_IP, 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("undefine 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("read %s failed\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 (1) + { + 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; + } + + close(clientSockfd); + if (serverRtpSockfd) + { + close(serverRtpSockfd); + } + if (serverRtcpSockfd > 0) + { + close(serverRtcpSockfd); + } + + free(rBuf); + free(sBuf); +} + +int main() +{ + + int rtspServerSockfd; + + int ret; + + rtspServerSockfd = createTcpSocket(); + if (rtspServerSockfd < 0) + { + printf("failed to create tcp socket\n"); + return -1; + } + + ret = bindSocketAddr(rtspServerSockfd, SERVER_IP, 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://%s:%d\n", __FILE__, SERVER_IP, 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); + } + + close(rtspServerSockfd); + + return 0; +} diff --git a/code/aac_rtsp/rtp.c b/code/aac_rtsp/rtp.c new file mode 100644 index 0000000000000000000000000000000000000000..b9670b45c7a11e1fd2bdcd4988f321965dd83d61 --- /dev/null +++ b/code/aac_rtsp/rtp.c @@ -0,0 +1,46 @@ +#include +#include +#include +#include +#include +#include +#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 rtpSendPacketOverUdp(int serverRtpSockfd, 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(serverRtpSockfd, (char *)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; +} diff --git a/code/aac_rtsp/rtp.h b/code/aac_rtsp/rtp.h new file mode 100644 index 0000000000000000000000000000000000000000..955f29fa0b8deda8320e510f83e741feaebf3fb0 --- /dev/null +++ b/code/aac_rtsp/rtp.h @@ -0,0 +1,70 @@ + +#include + +#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 | + * : .... : + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * + */ + +//定义RTP包头部的结构。它包含了一个12字节的头部，用来描述RTP数据包的一些信息 +//如版本号、负载类型、序列号、时间戳等 +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; +}; + +//定义了一个完整的RTP数据包的结构 +//包含了RtpHeader结构和一个长度不确定的负载部分 +struct RtpPacket +{ + struct RtpHeader rtpHeader; + uint8_t payload[0]; +}; + +//初始化RTP数据包的头部 +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); +//通过TCP协议发送RTP数据包 +//接受一个客户端的套接字描述符，一个RTP数据包的指针以及数据包的大小作为参数 +int rtpSendPacketOverTcp(int clientSockfd, struct RtpPacket *rtpPacket, uint32_t dataSize); + +//通过UDP协议额发送RTP数据包 +//接受一个服务器端的RTP套接字描述符、目标IP地址、目标端口号、RTP数据包的指针以及数据包的大小作为参数 +int rtpSendPacketOverUdp(int serverRtpSockfd, const char *ip, int16_t port, struct RtpPacket *rtpPacket, uint32_t dataSize); diff --git a/code/aac_rtsp/test.aac b/code/aac_rtsp/test.aac new file mode 100644 index 0000000000000000000000000000000000000000..47e13d4c9b3c538928dab04bca905d91cb682ebc Binary files /dev/null and b/code/aac_rtsp/test.aac differ