rtp的源码
一、RTP OV ER RTSP
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202/article/details/98986535 */ #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 | * : .... : * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * *版本号(V)：2Bit，用来标志使用RTP版本 *填充位§：1Bit，如果该位置位，则该RTP包的尾部就包含填充的附加字节 *扩展位(X)：1Bit，如果该位置位，则该RTP包的固定头部后面就跟着一个扩展头部 *CSRC技术器(CC)：4Bit，含有固定头部后面跟着的CSRC的数据 *标记位(M)：1Bit，该位的解释由配置文档来承担 *载荷类型(PT)：7Bit，标识了RTP载荷的类型 *序列号(SN)：16Bit，发送方在每发送完一个RTP包后就将该域的值增加1，可以由该域检测包的丢失及恢复 *包的序列。序列号的初始值是随机的 *时间戳：32比特，记录了该包中数据的第一个字节的采样时刻 *同步源标识符(SSRC)：32比特，同步源就是RTP包源的来源。在同一个RTP会话中不能有两个相同的SSRC值 *贡献源列表(CSRC List)：0-15项，每项32比特，这个不常用 */ 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二、RTP OVER UDP
/* * 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_
/* * 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; }
三、RTP中的数据为h264格式
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202/article/details/99089020 */ #include <stdio.h> #include <stdlib.h> #include <sys/types.h> #include <sys/socket.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <string.h> #include "rtp.h" #define H264_FILE_NAME "test.h264" #define CLIENT_IP "10.10.12.132" #define CLIENT_PORT 9832 #define FPS 25 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; } else { frameSize = (nextStartCode-frame); lseek(fd, frameSize-rSize, SEEK_CUR); } return frameSize; } static int createUdpSocket() { int fd; int on = 1; fd = socket(AF_INET, SOCK_DGRAM, 0); if(fd < 0) return -1; setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on)); return fd; } static int rtpSendH264Frame(int socket, char* ip, int16_t port, 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, ip, port, 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++) { //高三位：与NALU第一个字节的高三位相同 //Type：28，表示该RTP包一个分片 rtpPacket->payload[0] = (naluType & 0x60) | 28;//0110 0000 0001 1100 rtpPacket->payload[1] = naluType & 0x1F;//0001 1111 if (i == 0) //第一包数据 rtpPacket->payload[1] |= 0x80; // start 1000 0000 else if (remainPktSize == 0 && i == pktNum - 1) //最后一包数据 rtpPacket->payload[1] |= 0x40; // end 0100 0000 memcpy(rtpPacket->payload+2, frame+pos, RTP_MAX_PKT_SIZE); ret = rtpSendPacket(socket, ip, port, 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, ip, port, rtpPacket, remainPktSize+2); if(ret < 0) return -1; rtpPacket->rtpHeader.seq++; sendBytes += ret; } } out: return sendBytes; } int main(int argc, char* argv[]) { int socket; int fd; int fps = 25; int startCode; struct RtpPacket* rtpPacket;//udp的哪个字节 uint8_t* frame; uint32_t frameSize; fd = open(H264_FILE_NAME, O_RDONLY); if(fd < 0) { printf("failed to open %s\n", H264_FILE_NAME); return -1; } socket = createUdpSocket(); if(socket < 0) { printf("failed to create socket\n"); return -1; } rtpPacket = (struct RtpPacket*)malloc(500000); frame = (uint8_t*)malloc(500000); rtpHeaderInit(rtpPacket, 0, 0, 0, RTP_VESION, RTP_PAYLOAD_TYPE_H264, 0, 0, 0, 0x88923423); while(1) { frameSize = getFrameFromH264File(fd, frame, 500000); if(frameSize < 0) { printf("read err\n"); continue; } if(startCode3(frame)) startCode = 3; else startCode = 4; frameSize -= startCode; rtpSendH264Frame(socket, CLIENT_IP, CLIENT_PORT, rtpPacket, frame+startCode, frameSize); rtpPacket->rtpHeader.timestamp += 90000/FPS; usleep(1000*1000/fps); /* 假设时钟频率为90000，帧率为25 频率为90000表示一秒用90000点来表示 帧率为25，那么一帧就是1/25秒 所以一帧有90000*(1/25)=3600个点来表示 因此每一帧数据的时间增量为36008 */ } free(rtpPacket); free(frame); return 0; }
四、RTP中的数据为ACC格式
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202/article/details/99200935 */ #include <stdio.h> #include <stdlib.h> #include <sys/types.h> #include <sys/socket.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <string.h> #include "rtp.h" #define AAC_FILE "test.aac" #define CLIENT_PORT 9832 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; //2 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,剩2位 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，补2位,余5位 unsigned int adtsBufferFullness; //11 bit 0x7FF 说明是码率可变的码流 余2位 /* 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))//说明一个ADTS帧的开始 { res->id = ((unsigned int) in[1] & 0x08) >> 3;//0000 1000 printf("adts:id %d\n", res->id); res->layer = ((unsigned int) in[1] & 0x06) >> 1;//0000 0110 printf( "adts:layer %d\n", res->layer); res->protectionAbsent = (unsigned int) in[1] & 0x01; printf( "adts:protection_absent %d\n", res->protectionAbsent); res->profile = ((unsigned int) in[2] & 0xc0) >> 6; //1100 0000 printf( "adts:profile %d\n", res->profile); res->samplingFreqIndex = ((unsigned int) in[2] & 0x3c) >> 2;//0011 1100 printf( "adts:sf_index %d\n", res->samplingFreqIndex); res->privateBit = ((unsigned int) in[2] & 0x02) >> 1;//0000 0010 printf( "adts:pritvate_bit %d\n", res->privateBit); res->channelCfg = ((((unsigned int) in[2] & 0x01) << 2) | (((unsigned int) in[3] & 0xc0) >> 6));//1100 0000 printf( "adts:channel_configuration %d\n", res->channelCfg); res->originalCopy = ((unsigned int) in[3] & 0x20) >> 5;//0010 0000 printf( "adts:original %d\n", res->originalCopy); res->home = ((unsigned int) in[3] & 0x10) >> 4;////0001 0000 printf( "adts:home %d\n", res->home); res->copyrightIdentificationBit = ((unsigned int) in[3] & 0x08) >> 3;//0000 1000 printf( "adts:copyright_identification_bit %d\n", res->copyrightIdentificationBit); res->copyrightIdentificationStart = (unsigned int) in[3] & 0x04 >> 2;//0000 0100 printf( "adts:copyright_identification_start %d\n", res->copyrightIdentificationStart); res->aacFrameLength = (((((unsigned int) in[3]) & 0x03) << 11) | //0000 0011 (((unsigned int)in[4] & 0xFF) << 3) | ((unsigned int)in[5] & 0xE0) >> 5) ;//1110 0000 printf( "adts:aac_frame_length %d\n", res->aacFrameLength); res->adtsBufferFullness = (((unsigned int) in[5] & 0x1f) << 6 | //0001 1111 ((unsigned int) in[6] & 0xfc) >> 2);//1111 1100 printf( "adts:adts_buffer_fullness %d\n", res->adtsBufferFullness); res->numberOfRawDataBlockInFrame = ((unsigned int) in[6] & 0x03); printf( "adts:no_raw_data_blocks_in_frame %d\n", res->numberOfRawDataBlockInFrame); return 0; } else { printf("failed to parse adts header\n"); return -1; } } static int createUdpSocket() { int fd; int on = 1; fd = socket(AF_INET, SOCK_DGRAM, 0); if(fd < 0) return -1; setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on)); return fd; } static int rtpSendAACFrame(int socket, char* ip, int16_t port, struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize) { int ret; rtpPacket->payload[0] = 0x00;//前两位固定 rtpPacket->payload[1] = 0x10; rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; //高8位 111111110 0000 rtpPacket->payload[3] = (frameSize & 0x1F) << 3; //低5位 11111 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; } int main(int argc, char* argv[]) { int fd; int ret; int socket; uint8_t* frame; struct AdtsHeader adtsHeader; struct RtpPacket* rtpPacket; if(argc != 2) { printf("Usage: %s <dest ip>\n", argv[0]); return -1; } fd = open(AAC_FILE, O_RDONLY); if(fd < 0) { printf("failed to open %s\n", AAC_FILE); return -1; } socket = createUdpSocket(); if(socket < 0) { printf("failed to create udp socket\n"); return -1; } 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) { printf("--------------------------------\n"); ret = read(fd, frame, 7); if(ret <= 0) { lseek(fd, 0, SEEK_SET); continue; } 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(socket, argv[1], CLIENT_PORT, rtpPacket, frame, adtsHeader.aacFrameLength-7); usleep(23000); } close(fd); close(socket); free(frame); free(rtpPacket); return 0; }
五、一个传输H.264的RTSP服务器
从文件中赢取的。
rtp_h264.c六、一个传输ACC的RTSP服务器
从文件中赢取的。
/* * 作者：_JT_ * 博客：https://blog.csdn.net/weixin_42462202/article/details/99200935 */ #include <stdio.h> #include <stdlib.h> #include <sys/types.h> #include <sys/socket.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <string.h> #include "rtp.h" #define AAC_FILE "test.aac" #define CLIENT_PORT 9832 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; //2 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,剩2位 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，补2位,余5位 unsigned int adtsBufferFullness; //11 bit 0x7FF 说明是码率可变的码流 余2位 /* 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))//说明一个ADTS帧的开始 { res->id = ((unsigned int) in[1] & 0x08) >> 3;//0000 1000 printf("adts:id %d\n", res->id); res->layer = ((unsigned int) in[1] & 0x06) >> 1;//0000 0110 printf( "adts:layer %d\n", res->layer); res->protectionAbsent = (unsigned int) in[1] & 0x01; printf( "adts:protection_absent %d\n", res->protectionAbsent); res->profile = ((unsigned int) in[2] & 0xc0) >> 6; //1100 0000 printf( "adts:profile %d\n", res->profile); res->samplingFreqIndex = ((unsigned int) in[2] & 0x3c) >> 2;//0011 1100 printf( "adts:sf_index %d\n", res->samplingFreqIndex); res->privateBit = ((unsigned int) in[2] & 0x02) >> 1;//0000 0010 printf( "adts:pritvate_bit %d\n", res->privateBit); res->channelCfg = ((((unsigned int) in[2] & 0x01) << 2) | (((unsigned int) in[3] & 0xc0) >> 6));//1100 0000 printf( "adts:channel_configuration %d\n", res->channelCfg); res->originalCopy = ((unsigned int) in[3] & 0x20) >> 5;//0010 0000 printf( "adts:original %d\n", res->originalCopy); res->home = ((unsigned int) in[3] & 0x10) >> 4;////0001 0000 printf( "adts:home %d\n", res->home); res->copyrightIdentificationBit = ((unsigned int) in[3] & 0x08) >> 3;//0000 1000 printf( "adts:copyright_identification_bit %d\n", res->copyrightIdentificationBit); res->copyrightIdentificationStart = (unsigned int) in[3] & 0x04 >> 2;//0000 0100 printf( "adts:copyright_identification_start %d\n", res->copyrightIdentificationStart); res->aacFrameLength = (((((unsigned int) in[3]) & 0x03) << 11) | //0000 0011 (((unsigned int)in[4] & 0xFF) << 3) | ((unsigned int)in[5] & 0xE0) >> 5) ;//1110 0000 printf( "adts:aac_frame_length %d\n", res->aacFrameLength); res->adtsBufferFullness = (((unsigned int) in[5] & 0x1f) << 6 | //0001 1111 ((unsigned int) in[6] & 0xfc) >> 2);//1111 1100 printf( "adts:adts_buffer_fullness %d\n", res->adtsBufferFullness); res->numberOfRawDataBlockInFrame = ((unsigned int) in[6] & 0x03); printf( "adts:no_raw_data_blocks_in_frame %d\n", res->numberOfRawDataBlockInFrame); return 0; } else { printf("failed to parse adts header\n"); return -1; } } static int createUdpSocket() { int fd; int on = 1; fd = socket(AF_INET, SOCK_DGRAM, 0); if(fd < 0) return -1; setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, (const char*)&on, sizeof(on)); return fd; } static int rtpSendAACFrame(int socket, char* ip, int16_t port, struct RtpPacket* rtpPacket, uint8_t* frame, uint32_t frameSize) { int ret; rtpPacket->payload[0] = 0x00;//前两位固定 rtpPacket->payload[1] = 0x10; rtpPacket->payload[2] = (frameSize & 0x1FE0) >> 5; //高8位 111111110 0000 rtpPacket->payload[3] = (frameSize & 0x1F) << 3; //低5位 11111 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; } int main(int argc, char* argv[]) { int fd; int ret; int socket; uint8_t* frame; struct AdtsHeader adtsHeader; struct RtpPacket* rtpPacket; if(argc != 2) { printf("Usage: %s <dest ip>\n", argv[0]); return -1; } fd = open(AAC_FILE, O_RDONLY); if(fd < 0) { printf("failed to open %s\n", AAC_FILE); return -1; } socket = createUdpSocket(); if(socket < 0) { printf("failed to create udp socket\n"); return -1; } 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) { printf("--------------------------------\n"); ret = read(fd, frame, 7); if(ret <= 0) { lseek(fd, 0, SEEK_SET); continue; } 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(socket, argv[1], CLIENT_PORT, rtpPacket, frame, adtsHeader.aacFrameLength-7); usleep(23000); } close(fd); close(socket); free(frame); free(rtpPacket); return 0; }
