Add simulation of robust throughput estimator to the event log analyzer
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}
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index b8158f6..f33d96e 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -370,6 +370,7 @@
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_numerics",
"../rtc_base:stringutils",
+ "../test:explicit_key_value_config",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 8ca108e..6d84b1b 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
// 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));
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 974848b..9215b67 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -216,6 +216,20 @@
deps = [ "../system_wrappers:field_trial" ]
}
+rtc_library("explicit_key_value_config") {
+ sources = [
+ "explicit_key_value_config.cc",
+ "explicit_key_value_config.h",
+ ]
+
+ deps = [
+ "../api/transport:webrtc_key_value_config",
+ "../rtc_base:checks",
+ "../system_wrappers:field_trial",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
rtc_library("perf_test") {
visibility = [ "*" ]
testonly = true
diff --git a/test/explicit_key_value_config.cc b/test/explicit_key_value_config.cc
new file mode 100644
index 0000000..69f725a
--- /dev/null
+++ b/test/explicit_key_value_config.cc
@@ -0,0 +1,57 @@
+/*
+ * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/explicit_key_value_config.h"
+
+#include "api/transport/webrtc_key_value_config.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+ExplicitKeyValueConfig::ExplicitKeyValueConfig(const std::string& s) {
+ std::string::size_type field_start = 0;
+ while (field_start < s.size()) {
+ std::string::size_type separator_pos = s.find('/', field_start);
+ RTC_CHECK_NE(separator_pos, std::string::npos)
+ << "Missing separator '/' after field trial key.";
+ RTC_CHECK_GT(separator_pos, field_start)
+ << "Field trial key cannot be empty.";
+ std::string key = s.substr(field_start, separator_pos - field_start);
+ field_start = separator_pos + 1;
+
+ RTC_CHECK_LT(field_start, s.size())
+ << "Missing value after field trial key. String ended.";
+ separator_pos = s.find('/', field_start);
+ RTC_CHECK_NE(separator_pos, std::string::npos)
+ << "Missing terminating '/' in field trial string.";
+ RTC_CHECK_GT(separator_pos, field_start)
+ << "Field trial value cannot be empty.";
+ std::string value = s.substr(field_start, separator_pos - field_start);
+ field_start = separator_pos + 1;
+
+ key_value_map_[key] = value;
+ }
+ // This check is technically redundant due to earlier checks.
+ // We nevertheless keep the check to make it clear that the entire
+ // string has been processed, and without indexing past the end.
+ RTC_CHECK_EQ(field_start, s.size());
+}
+
+std::string ExplicitKeyValueConfig::Lookup(absl::string_view key) const {
+ auto it = key_value_map_.find(std::string(key));
+ if (it != key_value_map_.end())
+ return it->second;
+ return "";
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/test/explicit_key_value_config.h b/test/explicit_key_value_config.h
new file mode 100644
index 0000000..9a3bc84
--- /dev/null
+++ b/test/explicit_key_value_config.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
+#define TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/webrtc_key_value_config.h"
+
+namespace webrtc {
+namespace test {
+
+class ExplicitKeyValueConfig : public WebRtcKeyValueConfig {
+ public:
+ explicit ExplicitKeyValueConfig(const std::string& s);
+ std::string Lookup(absl::string_view key) const override;
+
+ private:
+ std::map<std::string, std::string> key_value_map_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
