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;
 }

上一篇                回目录