RTPSession *AudioStream::AddAudioWatcher(unsigned clientSessionId,char *sendAudioIp,int sendAudioPort,AudioCodec::RTPMap& rtpMap, AudioCodec::Type watcherCodec)
{
Log(">Add Audio Watcher[%u, %s,%d]\n",clientSessionId, sendAudioIp,sendAudioPort);
//Si tenemos video
if (sendAudioPort==0) {
Error("No video\n");
return NULL;
}
Rtps::iterator it = rtps.find(clientSessionId);
RTPSession* rtp = NULL;
//Si no esta
if (it == rtps.end()) {
Error("Rtp not init\n");
return NULL;
}
rtp = (*it).second;
if(!rtp->Init())
{
delete rtp;
Error("Rtp init error\n");
return NULL;
}
//Iniciamos las sesiones rtp de envio
if(!rtp->SetRemotePort(sendAudioIp,sendAudioPort))
{
delete rtp;
Error("Rtp SetRemotePort error\n");
return NULL;
}
rtp->SetSendingAudioRTPMap(rtpMap);
//Set video codec
if(!rtp->SetSendingAudioCodec(watcherCodec))
{
delete rtp;
//Error
Error("%s video codec not supported by peer\n",AudioCodec::GetNameFor(watcherCodec));
return NULL;
}
rtps[clientSessionId] = rtp;
//LOgeamos
Log("<Add Audio Watcher success [%u]\n",clientSessionId);
return rtp;
}
