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Add simulation of robust throughput estimator to the event log analyzer

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Bug: webrtc:11566
Change-Id: I873d1c1bd6682a973b3a130289390e09ef47cc37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177017
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31538}
This commit is contained in:
Björn Terelius
2020-06-17 11:55:55 +02:00
committed by Commit Bot
parent a46fce2194
commit ae1892d4e4
5 changed files with 155 additions and 31 deletions
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79
rtc_tools/rtc_event_log_visualizer/analyzer.cc
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@ -56,10 +56,7 @@
#include "rtc_base/rate_statistics.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
#ifndef BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
#define BWE_TEST_LOGGING_COMPILE_TIME_ENABLE 0
#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
#include "test/explicit_key_value_config.h"
namespace webrtc {
@ -1212,10 +1209,13 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
PointStyle::kHighlight);
TimeSeries acked_time_series("Acked bitrate", LineStyle::kLine,
TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine,
PointStyle::kHighlight);
TimeSeries acked_estimate_time_series(
"Acked bitrate estimate", LineStyle::kLine, PointStyle::kHighlight);
TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine,
PointStyle::kHighlight);
TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate",
LineStyle::kLine,
PointStyle::kHighlight);
auto rtp_iterator = outgoing_rtp.begin();
auto rtcp_iterator = incoming_rtcp.begin();
@ -1241,20 +1241,18 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
return std::numeric_limits<int64_t>::max();
};
RateStatistics acked_bitrate(250, 8000);
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
FieldTrialBasedConfig field_trial_config_;
// The event_log_visualizer should normally not be compiled with
// BWE_TEST_LOGGING_COMPILE_TIME_ENABLE since the normal plots won't work.
// However, compiling with BWE_TEST_LOGGING, running with --plot=sendside_bwe
// and piping the output to plot_dynamics.py can be used as a hack to get the
// internal state of various BWE components. In this case, it is important
// we don't instantiate the AcknowledgedBitrateEstimator both here and in
// GoogCcNetworkController since that would lead to duplicate outputs.
RateStatistics 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/");
std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
robust_throughput_estimator(
AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
FieldTrialBasedConfig field_trial_config;
std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
acknowledged_bitrate_estimator(
AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_));
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config));
int64_t time_us =
std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
int64_t last_update_us = 0;
@ -1264,24 +1262,40 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
RtpPacketSendInfo packet_info;
packet_info.ssrc = rtp_packet.rtp.header.ssrc;
packet_info.transport_sequence_number =
rtp_packet.rtp.header.extension.transportSequenceNumber;
packet_info.rtp_sequence_number = rtp_packet.rtp.header.sequenceNumber;
packet_info.length = rtp_packet.rtp.total_length;
if (IsRtxSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
rtp_packet.rtp.header.ssrc)) {
// Don't set the optional media type as we don't know if it is
// a retransmission, FEC or padding.
} else if (IsVideoSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
rtp_packet.rtp.header.ssrc)) {
packet_info.packet_type = RtpPacketMediaType::kVideo;
} else if (IsAudioSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
rtp_packet.rtp.header.ssrc)) {
packet_info.packet_type = RtpPacketMediaType::kAudio;
}
transport_feedback.AddPacket(
packet_info,
0u, // Per packet overhead bytes.
Timestamp::Micros(rtp_packet.rtp.log_time_us()));
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
if (sent_msg)
observer.Update(goog_cc->OnSentPacket(*sent_msg));
}
rtc::SentPacket sent_packet;
sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
sent_packet.info.included_in_allocation = true;
sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
sent_packet.packet_id =
rtp_packet.rtp.header.extension.transportSequenceNumber;
sent_packet.info.included_in_feedback = true;
}
auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
if (sent_msg)
observer.Update(goog_cc->OnSentPacket(*sent_msg));
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
@ -1296,13 +1310,13 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
std::vector<PacketResult> feedback =
feedback_msg->SortedByReceiveTime();
if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
feedback);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
for (const PacketResult& packet : feedback)
robust_throughput_estimator->IncomingPacketFeedbackVector(feedback);
for (const PacketResult& packet : feedback) {
acked_bitrate.Update(packet.sent_packet.size.bytes(),
packet.receive_time.ms());
}
bitrate_bps = acked_bitrate.Rate(feedback.back().receive_time.ms());
}
}
@ -1310,12 +1324,14 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
float x = config_.GetCallTimeSec(clock.TimeInMicroseconds());
float y = bitrate_bps.value_or(0) / 1000;
acked_time_series.points.emplace_back(x, y);
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
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);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {
@ -1336,6 +1352,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
}
// Add the data set to the plot.
plot->AppendTimeSeries(std::move(time_series));
plot->AppendTimeSeries(std::move(robust_time_series));
plot->AppendTimeSeries(std::move(acked_time_series));
plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));