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Improved loss handling for bitrate reductions in WebRTC push input

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This commit is contained in:
Thulinma 2020-06-02 15:32:20 +02:00
parent 7a5f5d5855
commit 78a1cd240d
2 changed files with 41 additions and 11 deletions
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51
src/output/output_webrtc.cpp
View file

@ -1642,23 +1642,52 @@ namespace Mist{
stats_lossperc = (double)(rtcTrack.sorter.lostCurrent * 100.) / (double)(rtcTrack.sorter.lostCurrent + rtcTrack.sorter.packCurrent);
stats_jitter = rtcTrack.jitter/rtcTrack.rtpToDTSC.multiplier;
stats_lossnum = rtcTrack.sorter.lostTotal;
//If we have > 5% loss, constrain video by 10%
if (stats_lossperc > 5){
videoConstraint *= 0.9;
if (videoConstraint < 1024){videoConstraint = 1024;}
JSON::Value commandResult;
commandResult["type"] = "on_video_bitrate";
commandResult["result"] = true;
commandResult["video_bitrate"] = videoBitrate;
commandResult["video_bitrate_constraint"] = videoConstraint;
webSock->sendFrame(commandResult.toString());
}
//Print stats at appropriate log levels
if (stats_lossperc > 1 || stats_jitter > 20){
INFO_MSG("Receiver Report (%s): %.2f%% loss, %" PRIu32 " total lost, %.2f ms jitter", rtcTrack.rtpToDTSC.codec.c_str(), stats_lossperc, rtcTrack.sorter.lostTotal, stats_jitter);
}else{
HIGH_MSG("Receiver Report (%s): %.2f%% loss, %" PRIu32 " total lost, %.2f ms jitter", rtcTrack.rtpToDTSC.codec.c_str(), stats_lossperc, rtcTrack.sorter.lostTotal, stats_jitter);
}
//Calculate loss percentage average over moving window
stats_loss_avg.push_back(stats_lossperc);
while (stats_loss_avg.size() > 5){stats_loss_avg.pop_front();}
if (stats_loss_avg.size()){
double curr_avg_loss = 0;
for (std::deque<double>::iterator it = stats_loss_avg.begin(); it != stats_loss_avg.end(); ++it){
curr_avg_loss += *it;
}
curr_avg_loss /= stats_loss_avg.size();
uint32_t preConstraint = videoConstraint;
if (curr_avg_loss > 50){
//If we have > 50% loss, constrain video by 9%
videoConstraint *= 0.91;
}else if (curr_avg_loss > 10){
//If we have > 10% loss, constrain video by 5%
videoConstraint *= 0.95;
}else if (curr_avg_loss > 5){
//If we have > 5% loss, constrain video by 1%
videoConstraint *= 0.99;
}
//Do not reduce under 32 kbps
if (videoConstraint < 1024*32){videoConstraint = 1024*32;}
if (videoConstraint != preConstraint){
INFO_MSG("Reduced video bandwidth maximum to %" PRIu32 " because average loss is %.2f", videoConstraint, curr_avg_loss);
JSON::Value commandResult;
commandResult["type"] = "on_video_bitrate";
commandResult["result"] = true;
commandResult["video_bitrate"] = videoBitrate;
commandResult["video_bitrate_constraint"] = videoConstraint;
webSock->sendFrame(commandResult.toString());
}
}
if (packetLog.is_open()){
packetLog << "[" << Util::bootMS() << "] Receiver Report (" << rtcTrack.rtpToDTSC.codec << "): " << stats_lossperc << " percent loss, " << rtcTrack.sorter.lostTotal << " total lost, " << stats_jitter << " ms jitter" << std::endl;
}

1
src/output/output_webrtc.h
View file

@ -220,6 +220,7 @@ namespace Mist{
uint64_t stats_nacknum;
uint64_t stats_lossnum;
double stats_lossperc;
std::deque<double> stats_loss_avg;
#if defined(WEBRTC_PCAP)
PCAPWriter pcapOut; ///< Used during development to write unprotected packets that can be