void MediaStream::setTransportInfo(std::string audio_info, std::string video_info) {
if (video_enabled_) {
uint32_t video_sink_ssrc = getVideoSinkSSRC();
uint32_t video_source_ssrc = getVideoSourceSSRC();
if (video_sink_ssrc != kDefaultVideoSinkSSRC) {
stats_->getNode()[video_sink_ssrc].insertStat("clientHostType", StringStat{video_info});
}
if (video_source_ssrc != 0) {
stats_->getNode()[video_source_ssrc].insertStat("clientHostType", StringStat{video_info});
}
}
if (audio_enabled_) {
uint32_t audio_sink_ssrc = getAudioSinkSSRC();
uint32_t audio_source_ssrc = getAudioSourceSSRC();
if (audio_sink_ssrc != kDefaultAudioSinkSSRC) {
stats_->getNode()[audio_sink_ssrc].insertStat("clientHostType", StringStat{audio_info});
}
if (audio_source_ssrc != 0) {
stats_->getNode()[audio_source_ssrc].insertStat("clientHostType", StringStat{audio_info});
}
}
}
void MediaStream::read(std::shared_ptr<DataPacket> packet) {
char* buf = packet->data;
int len = packet->length;
// PROCESS RTCP
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
uint32_t recvSSRC = 0;
if (!chead->isRtcp()) {
recvSSRC = head->getSSRC();
} else if (chead->packettype == RTCP_Sender_PT) { // Sender Report
recvSSRC = chead->getSSRC();
}
// DELIVER FEEDBACK (RR, FEEDBACK PACKETS)
if (chead->isFeedback()) {
if (fb_sink_ != nullptr && should_send_feedback_) {
fb_sink_->deliverFeedback(std::move(packet));
}
} else {
// RTP or RTCP Sender Report
if (bundle_) {
// Check incoming SSRC
// Deliver data
if (isVideoSourceSSRC(recvSSRC)) {
parseIncomingPayloadType(buf, len, VIDEO_PACKET);
video_sink_->deliverVideoData(std::move(packet));
} else if (isAudioSourceSSRC(recvSSRC)) {
parseIncomingPayloadType(buf, len, AUDIO_PACKET);
audio_sink_->deliverAudioData(std::move(packet));
} else {
ELOG_DEBUG("%s read video unknownSSRC: %u, localVideoSSRC: %u, localAudioSSRC: %u",
toLog(), recvSSRC, this->getVideoSourceSSRC(), this->getAudioSourceSSRC());
}
} else {
if (packet->type == AUDIO_PACKET && audio_sink_ != nullptr) {
parseIncomingPayloadType(buf, len, AUDIO_PACKET);
// Firefox does not send SSRC in SDP
if (getAudioSourceSSRC() == 0) {
ELOG_DEBUG("%s discoveredAudioSourceSSRC:%u", toLog(), recvSSRC);
this->setAudioSourceSSRC(recvSSRC);
}
audio_sink_->deliverAudioData(std::move(packet));
} else if (packet->type == VIDEO_PACKET && video_sink_ != nullptr) {
parseIncomingPayloadType(buf, len, VIDEO_PACKET);
// Firefox does not send SSRC in SDP
if (getVideoSourceSSRC() == 0) {
ELOG_DEBUG("%s discoveredVideoSourceSSRC:%u", toLog(), recvSSRC);
this->setVideoSourceSSRC(recvSSRC);
}
// change ssrc for RTP packets, don't touch here if RTCP
video_sink_->deliverVideoData(std::move(packet));
}
} // if not bundle
} // if not Feedback
}
