一个基于JRTPLIB的轻量级RTSP客户端(myRTSPClient)——实现篇：（十）使用JRTPLIB传输RTP数据

myRtspClient通过简单修改JRTPLIB的官方例程作为其RTP传输层实现。因为JRTPLIB使用的是CMAKE编译工具，这就是为什么编译myRtspClient时需要预装CMAKE。

该部分所有代码均集中在myRtpSession.cpp中，接下来将对其进行分析。

一、获取RTP数据

此处GetMyRTPData获取数据的方式主要是轮询，即每隔USLEEP_UNIT个微秒轮询一次直到获取到一包数据或超时，超时时间为timeout_ms，单位是微秒。

GetMyRTPPacket的逻辑与之相同，但是会比前者多获取RTP消息头，对于myRtspClient的用户来说，这同样也就是RtspClient::GetMediaPacket和RtspClient::GetMediaData的区别（逻辑相同，GetMyRTPPacket的代码就不再此附上了）。

RtspClient::GetMediaPacket和RtspClient::GetMediaData最终获取RTP数据就是通过调用这两个函数完成的。

 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 - ) / 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 = ;

         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 = ; // Got the data. So not need to sleep any more.

     } while(UsleepTimes > );

     return data_buf;

 }

二、会话建立、结束等接口

此处的MyRTP_SetUp的作用是建立会话，它会确定RTP/RTCP的UDP端口，建立通信socket。每当RTSP的SETUP命令设置成功后，都会调用此函数。

 int MyRTPSession::MyRTP_SetUp(MediaSession * media_session)

 {

     if(!media_session) {

         fprintf(stderr, "%s: Invalid media session\n", __func__);

         return RTP_ERROR;

     }

     if( == media_session->TimeRate) {

         fprintf(stderr, "%s: Invalid MediaSession::TimeRate\n", __func__);

         return RTP_ERROR;

     }

     if( == 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);

 }

客户端通过MyRTP_Teardown发起销毁会话，每当RTSP的TEARDOWN命令设置成功后，都会调用此函数。

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

 {

     struct timeval Timeout;

     if(!tval) {

         Timeout.tv_sec = ;

         Timeout.tv_usec = ;

     } else {

         Timeout.tv_sec = tval->tv_sec;

         Timeout.tv_usec = tval->tv_usec;

     }

     media_session->RTPPort = ;

     BYEDestroy(RTPTime(Timeout.tv_sec, Timeout.tv_usec), , );

 }

客户端通过OnBYEPacket被动销毁会话，当服务器向客户端发送BYE的RTP数据包时（比如当一段媒体流播放完的时候），该函数就会被调用。其中DestroiedClbk是myRtspClient提供给用户的回调接口。用户可以通过调用RtspClient::SetAudioByeFromServerClbk/RtspClient::SetVideoByeFromServerClbk来设置该函数。（逻辑相同，OnRemoveSource的代码就不再此附上了）。

 void MyRTPSession::OnBYEPacket(RTPSourceData *dat)

 {

     if (dat->IsOwnSSRC())

         return;

     uint32_t ip;

     uint16_t port;

     if (dat->GetRTPDataAddress() != )

     {

         const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTPDataAddress());

         ip = addr->GetIP();

         port = addr->GetPort();

     }

     else if (dat->GetRTCPDataAddress() != )

     {

         const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTCPDataAddress());

         ip = addr->GetIP();

         port = addr->GetPort()-;

     }

     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();

     }

 }

每当新加入一个RTP数据源，OnNewSource就会被调用

 void MyRTPSession::OnNewSource(RTPSourceData *dat)

 {

     if (dat->IsOwnSSRC())

         return;

     uint32_t ip;

     uint16_t port;

     if (dat->GetRTPDataAddress() != )

     {

         const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTPDataAddress());

         ip = addr->GetIP();

         port = addr->GetPort();

     }

     else if (dat->GetRTCPDataAddress() != )

     {

         const RTPIPv4Address *addr = (const RTPIPv4Address *)(dat->GetRTCPDataAddress());

         ip = addr->GetIP();

         port = addr->GetPort()-;

     }

     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;

 }

巴特西

一个基于JRTPLIB的轻量级RTSP客户端(myRTSPClient)——实现篇：（十）使用JRTPLIB传输RTP数据

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