rtp的源码

一、RTP OV ER RTSP 

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/*
 * 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.h
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tcp_rtp.c

二、RTP  OVER UDP

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/*
 * 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.h
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/*
 * 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;
}
rtp.c
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 三、RTP中的数据为h264格式

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/*
 * 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;
}
rtp_h264.c
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 四、RTP中的数据为ACC格式

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/*
 * 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;
}
View Code
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 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。

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/*
 * 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;
}
rtp_aac.c
复制代码

 

posted @   泽良_小涛  阅读(692)  评论(0)  编辑  收藏  举报
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