一個基於JRTPLIB的輕量級RTSP客戶端(myRTSPClient)——實現篇：（十）使用JRTPLIB傳輸RTP數據

myRtspClient通過簡單修改JRTPLIB的官方例程作為其RTP傳輸層實現。因為JRTPLIB使用的是CMAKE編譯工具，這就是為什么編譯myRtspClient時需要預裝CMAKE。

該部分所有代碼均集中在myRtpSession.cpp中，接下來將對其進行分析。

 

一、獲取RTP數據

此處GetMyRTPData獲取數據的方式主要是輪詢，即每隔USLEEP_UNIT個微秒輪詢一次直到獲取到一包數據或超時，超時時間為timeout_ms，單位是微秒。

GetMyRTPPacket的邏輯與之相同，但是會比前者多獲取RTP消息頭，對於myRtspClient的用戶來說，這同樣也就是RtspClient::GetMediaPacket和RtspClient::GetMediaData的區別（邏輯相同，GetMyRTPPacket的代碼就不再此附上了）。

RtspClient::GetMediaPacket和RtspClient::GetMediaData最終獲取RTP數據就是通過調用這兩個函數完成的。

uint8_t * MyRTPSession::GetMyRTPData(uint8_t * data_buf, size_t * size, unsigned long timeout_ms)
{
    if(!data_buf) {
        fprintf(stderr, "%s: Invalide argument('data_buf==NULL')", __func__);
        return NULL;
    }

    if(!size) {
        fprintf(stderr, "%s: Invalide argument('size==NULL')", __func__);
        return NULL;
    }

    unsigned long UsleepTimes = (timeout_ms + USLEEP_UNIT - 1) / USLEEP_UNIT; // floor the 'timeout_ms / USLEEP_UNIT'

    do {
#ifndef RTP_SUPPORT_THREAD
        int status = Poll();
        if(!IsError(status)) return NULL;
#endif 

        BeginDataAccess();

        // check incoming packets
        if (!GotoFirstSourceWithData()) {
            EndDataAccess();
            usleep(USLEEP_UNIT);
            UsleepTimes--;
            continue;
            // return NULL;
        }
        RTPPacket *pack;

        if(!(pack = GetNextPacket()))
        {
            EndDataAccess();
            usleep(USLEEP_UNIT);
            UsleepTimes--;
            continue;
            // return NULL;
        }

        size_t PacketSize = 0;
        uint8_t * Packet = NULL;
        Packet = pack->GetPayloadData();
        PacketSize = pack->GetPayloadLength();
        // printf("data length: %lu\n", PacketSize);

        *size = PacketSize;
        memcpy(data_buf, Packet, PacketSize);

        // we don't longer need the packet, so
        // we'll delete it
        DeletePacket(pack);
        EndDataAccess();
        UsleepTimes = 0; // Got the data. So not need to sleep any more.
    } while(UsleepTimes > 0);

    return data_buf;
}

 

二、會話建立、結束等接口

此處的MyRTP_SetUp的作用是建立會話，它會確定RTP/RTCP的UDP端口，建立通信socket。每當RTSP的SETUP命令設置成功后，都會調用此函數。

int MyRTPSession::MyRTP_SetUp(MediaSession * media_session)
{
    if(!media_session) {
        fprintf(stderr, "%s: Invalid media session\n", __func__);
        return RTP_ERROR;
    }
    if(0 == media_session->TimeRate) {
        fprintf(stderr, "%s: Invalid MediaSession::TimeRate\n", __func__);
        return RTP_ERROR;
    }
    if(0 == media_session->RTPPort) {
        fprintf(stderr, "%s: Invalid MediaSession::RTPPort\n", __func__);
        return RTP_ERROR;
    }

    int status;

    // Now, we'll create a RTP session, set the destination
    // and poll for incoming data.

    RTPUDPv4TransmissionParams transparams;
    RTPSessionParams sessparams;

    // IMPORTANT: The local timestamp unit MUST be set, otherwise
    //            RTCP Sender Report info will be calculated wrong
    // In this case, we'll be just use 8000 samples per second.
    sessparams.SetOwnTimestampUnit(1.0/media_session->TimeRate);         

    sessparams.SetAcceptOwnPackets(true);
    transparams.SetPortbase(media_session->RTPPort);
    status = Create(sessparams,&transparams);  
    return IsError(status);
}

 

客戶端通過MyRTP_Teardown發起銷毀會話，每當RTSP的TEARDOWN命令設置成功后，都會調用此函數。

void MyRTPSession::MyRTP_Teardown(MediaSession * media_session, struct timeval * tval)
{
    struct timeval Timeout;

    if(!tval) {
        Timeout.tv_sec = 1; 
        Timeout.tv_usec = 0; 
    } else {
        Timeout.tv_sec = tval->tv_sec;
        Timeout.tv_usec = tval->tv_usec;
    }

    media_session->RTPPort = 0;
    BYEDestroy(RTPTime(Timeout.tv_sec, Timeout.tv_usec), 0, 0);
}

 

客戶端通過OnBYEPacket被動銷毀會話，當服務器向客戶端發送BYE的RTP數據包時（比如當一段媒體流播放完的時候），該函數就會被調用。其中DestroiedClbk是myRtspClient提供給用戶的回調接口。用戶可以通過調用RtspClient::SetAudioByeFromServerClbk/RtspClient::SetVideoByeFromServerClbk來設置該函數。（邏輯相同，OnRemoveSource的代碼就不再此附上了）。

void MyRTPSession::OnBYEPacket(RTPSourceData *dat)
{
    if (dat->IsOwnSSRC())
        return;

    uint32_t ip;
    uint16_t port;

    if (dat->GetRTPDataAddress() != 0)
    {
        const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTPDataAddress());
        ip = addr->GetIP();
        port = addr->GetPort();
    }
    else if (dat->GetRTCPDataAddress() != 0)
    {
        const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTCPDataAddress());
        ip = addr->GetIP();
        port = addr->GetPort()-1;
    }
    else
        return;

    RTPIPv4Address dest(ip,port);
    DeleteDestination(dest);

    struct in_addr inaddr;
    inaddr.s_addr = htonl(ip);
    std::cout << "Deleting destination " << std::string(inet_ntoa(inaddr)) << ":" << port << std::endl;
    if(DestroiedClbk) {
        DestroiedClbk();
    } 
}

 

每當新加入一個RTP數據源，OnNewSource就會被調用

void MyRTPSession::OnNewSource(RTPSourceData *dat)
{
    if (dat->IsOwnSSRC())
        return;

    uint32_t ip;
    uint16_t port;

    if (dat->GetRTPDataAddress() != 0)
    {
        const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTPDataAddress());
        ip = addr->GetIP();
        port = addr->GetPort();
    }
    else if (dat->GetRTCPDataAddress() != 0)
    {
        const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTCPDataAddress());
        ip = addr->GetIP();
        port = addr->GetPort()-1;
    }
    else
        return;

    RTPIPv4Address dest(ip,port);
    AddDestination(dest);

    struct in_addr inaddr;
    inaddr.s_addr = htonl(ip);
    std::cout << "Adding destination " << std::string(inet_ntoa(inaddr)) << ":" << port << std::endl;
}

 

 

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