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 - 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、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(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 = 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() != 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() != 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;
}