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 ExternalOutput::queueData(char* buffer, int length, packetType type) {
if (!recording_) {
return;
}
RtcpHeader *head = reinterpret_cast<RtcpHeader*>(buffer);
if (head->isRtcp()) {
return;
}
if (first_data_received_ == time_point()) {
first_data_received_ = clock::now();
if (getAudioSinkSSRC() == 0) {
ELOG_DEBUG("No audio detected");
audio_map_ = RtpMap{0, "PCMU", 8000, AUDIO_TYPE, 1};
audio_codec_ = AV_CODEC_ID_PCM_MULAW;
}
}
if (need_to_send_fir_ && video_source_ssrc_) {
sendFirPacket();
need_to_send_fir_ = false;
}
if (type == VIDEO_PACKET) {
RtpHeader* h = reinterpret_cast<RtpHeader*>(buffer);
uint8_t payloadtype = h->getPayloadType();
if (video_offset_ms_ == -1) {
video_offset_ms_ = ClockUtils::durationToMs(clock::now() - first_data_received_);
ELOG_DEBUG("File %s, video offset msec: %llu", context_->filename, video_offset_ms_);
video_queue_.setTimebase(video_maps_[payloadtype].clock_rate);
}
// If this is a red header, let's push it to our fec_receiver_, which will spit out frames in one
// of our other callbacks.
// Otherwise, just stick it straight into the video queue.
if (payloadtype == RED_90000_PT) {
// The only things AddReceivedRedPacket uses are headerLength and sequenceNumber.
// Unfortunately the amount of crap
// we would have to pull in from the WebRtc project to fully construct
// a webrtc::RTPHeader object is obscene. So
// let's just do this hacky fix.
webrtc::RTPHeader hacky_header;
hacky_header.headerLength = h->getHeaderLength();
hacky_header.sequenceNumber = h->getSeqNumber();
// AddReceivedRedPacket returns 0 if there's data to process
if (0 == fec_receiver_->AddReceivedRedPacket(hacky_header, (const uint8_t*)buffer,
length, ULP_90000_PT)) {
fec_receiver_->ProcessReceivedFec();
}
} else {
video_queue_.pushPacket(buffer, length);
}
} else {
if (audio_offset_ms_ == -1) {
audio_offset_ms_ = ClockUtils::durationToMs(clock::now() - first_data_received_);
ELOG_DEBUG("File %s, audio offset msec: %llu", context_->filename, audio_offset_ms_);
// Let's also take a moment to set our audio queue timebase.
RtpHeader* h = reinterpret_cast<RtpHeader*>(buffer);
if (h->getPayloadType() == PCMU_8000_PT) {
audio_queue_.setTimebase(8000);
} else if (h->getPayloadType() == OPUS_48000_PT) {
audio_queue_.setTimebase(48000);
}
}
audio_queue_.pushPacket(buffer, length);
}
if (audio_queue_.hasData() || video_queue_.hasData()) {
// One or both of our queues has enough data to write stuff out. Notify our writer.
cond_.notify_one();
}
}
