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Improved RobustThroughputEstimator

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- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed).
- Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.)
- Add tests cases for high loss, deep reordering and paused/resumed streams to unittest.
- Remove some field trial settings that have very minor effect and rename some of the others.
- Change analyzer.cc to only draw data points if the estimators have valid estimates.

Bug: webrtc:13402
Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36849}
This commit is contained in:
Björn Terelius
2022-05-05 15:54:49 +02:00
committed by WebRTC LUCI CQ
parent 2cdbb969f0
commit eb9af84a55
7 changed files with 592 additions and 248 deletions
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43
rtc_tools/rtc_event_log_visualizer/analyzer.cc
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@ -1242,11 +1242,11 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
return std::numeric_limits<int64_t>::max();
};
RateStatistics acked_bitrate(750, 8000);
RateStatistics raw_acked_bitrate(750, 8000);
test::ExplicitKeyValueConfig throughput_config(
"WebRTC-Bwe-RobustThroughputEstimatorSettings/"
"enabled:true,reduce_bias:true,assume_shared_link:false,initial_packets:"
"10,min_packets:25,window_duration:750ms,unacked_weight:0.5/");
"enabled:true,required_packets:10,"
"window_packets:25,window_duration:1000ms,unacked_weight:1.0/");
std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
robust_throughput_estimator(
AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
@ -1305,7 +1305,6 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
auto feedback_msg = transport_feedback.ProcessTransportFeedback(
rtcp_iterator->transport_feedback,
Timestamp::Millis(clock.TimeInMilliseconds()));
absl::optional<uint32_t> bitrate_bps;
if (feedback_msg) {
observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
std::vector<PacketResult> feedback =
@ -1315,24 +1314,30 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
feedback);
robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
for (const PacketResult& packet : feedback) {
acked_bitrate.Update(packet.sent_packet.size.bytes(),
packet.receive_time.ms());
raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
packet.receive_time.ms());
}
absl::optional<uint32_t> raw_bitrate_bps =
raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
float x = config_.GetCallTimeSec(clock.CurrentTime());
if (raw_bitrate_bps) {
float y = raw_bitrate_bps.value() / 1000;
acked_time_series.points.emplace_back(x, y);
}
absl::optional<DataRate> robust_estimate =
robust_throughput_estimator->bitrate();
if (robust_estimate) {
float y = robust_estimate.value().kbps();
robust_time_series.points.emplace_back(x, y);
}
absl::optional<DataRate> acked_estimate =
acknowledged_bitrate_estimator->bitrate();
if (acked_estimate) {
float y = acked_estimate.value().kbps();
acked_estimate_time_series.points.emplace_back(x, y);
}
bitrate_bps = acked_bitrate.Rate(feedback.back().receive_time.ms());
}
}
float x = config_.GetCallTimeSec(clock.CurrentTime());
float y = bitrate_bps.value_or(0) / 1000;
acked_time_series.points.emplace_back(x, y);
y = robust_throughput_estimator->bitrate()
.value_or(DataRate::Zero())
.kbps();
robust_time_series.points.emplace_back(x, y);
y = acknowledged_bitrate_estimator->bitrate()
.value_or(DataRate::Zero())
.kbps();
acked_estimate_time_series.points.emplace_back(x, y);
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {