Add simulation of receive-side bandwidth estimate to event_log_analyzer.
Previously reviewed at https://codereview.webrtc.org/2986683002/
Bug: webrtc:7726
Change-Id: I9568bd8387d79f313d6c7d53ded7c23460df1598
Reviewed-on: https://webrtc-review.googlesource.com/6360
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20141}diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index af38964..173106a 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -38,6 +38,7 @@
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
+#include "rtc_base/logging.h"
namespace {
@@ -386,6 +387,7 @@
RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
+ bool default_map_used = false;
webrtc::ParsedRtcEventLog parsed_stream;
if (!parsed_stream.ParseFile(input_file)) {
@@ -433,8 +435,12 @@
parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
&total_length, nullptr);
- if (extension_map == nullptr)
+ if (extension_map == nullptr) {
extension_map = &default_map;
+ if (!default_map_used)
+ LOG(LS_WARNING) << "Using default header extension map";
+ default_map_used = true;
+ }
// Parse header to get SSRC and RTP time.
webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 9447483..6c42a72 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -30,6 +30,7 @@
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "modules/congestion_controller/include/send_side_congestion_controller.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -1107,7 +1108,7 @@
}
}
-void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) {
+void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
std::multimap<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
std::multimap<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
@@ -1226,7 +1227,85 @@
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
- plot->SetTitle("Simulated BWE behavior");
+ plot->SetTitle("Simulated send-side BWE behavior");
+}
+
+void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
+ class RembInterceptingPacketRouter : public PacketRouter {
+ public:
+ void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+ uint32_t bitrate_bps) override {
+ last_bitrate_bps_ = bitrate_bps;
+ bitrate_updated_ = true;
+ PacketRouter::OnReceiveBitrateChanged(ssrcs, bitrate_bps);
+ }
+ uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
+ bool GetAndResetBitrateUpdated() {
+ bool bitrate_updated = bitrate_updated_;
+ bitrate_updated_ = false;
+ return bitrate_updated;
+ }
+
+ private:
+ uint32_t last_bitrate_bps_;
+ bool bitrate_updated_;
+ };
+
+ std::multimap<uint64_t, const LoggedRtpPacket*> incoming_rtp;
+
+ for (const auto& kv : rtp_packets_) {
+ if (kv.first.GetDirection() == PacketDirection::kIncomingPacket &&
+ IsVideoSsrc(kv.first)) {
+ for (const LoggedRtpPacket& rtp_packet : kv.second)
+ incoming_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
+ }
+ }
+
+ SimulatedClock clock(0);
+ RembInterceptingPacketRouter packet_router;
+ // TODO(terelius): The PacketRrouter is the used as the RemoteBitrateObserver.
+ // Is this intentional?
+ ReceiveSideCongestionController rscc(&clock, &packet_router);
+ // TODO(holmer): Log the call config and use that here instead.
+ // static const uint32_t kDefaultStartBitrateBps = 300000;
+ // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
+
+ TimeSeries time_series("Receive side estimate", LINE_DOT_GRAPH);
+ TimeSeries acked_time_series("Received bitrate", LINE_GRAPH);
+
+ RateStatistics acked_bitrate(250, 8000);
+ int64_t last_update_us = 0;
+ for (const auto& kv : incoming_rtp) {
+ const LoggedRtpPacket& packet = *kv.second;
+ int64_t arrival_time_ms = packet.timestamp / 1000;
+ size_t payload = packet.total_length; /*Should subtract header?*/
+ clock.AdvanceTimeMicroseconds(packet.timestamp -
+ clock.TimeInMicroseconds());
+ rscc.OnReceivedPacket(arrival_time_ms, payload, packet.header);
+ acked_bitrate.Update(payload, arrival_time_ms);
+ rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+ if (bitrate_bps) {
+ uint32_t y = *bitrate_bps / 1000;
+ float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
+ 1000000;
+ acked_time_series.points.emplace_back(x, y);
+ }
+ if (packet_router.GetAndResetBitrateUpdated() ||
+ clock.TimeInMicroseconds() - last_update_us >= 1e6) {
+ uint32_t y = packet_router.last_bitrate_bps() / 1000;
+ float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
+ 1000000;
+ time_series.points.emplace_back(x, y);
+ last_update_us = clock.TimeInMicroseconds();
+ }
+ }
+ // Add the data set to the plot.
+ plot->AppendTimeSeries(std::move(time_series));
+ plot->AppendTimeSeries(std::move(acked_time_series));
+
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
+ plot->SetTitle("Simulated receive-side BWE behavior");
}
void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
diff --git a/rtc_tools/event_log_visualizer/analyzer.h b/rtc_tools/event_log_visualizer/analyzer.h
index 65ca19bf..5504282 100644
--- a/rtc_tools/event_log_visualizer/analyzer.h
+++ b/rtc_tools/event_log_visualizer/analyzer.h
@@ -90,7 +90,8 @@
void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
- void CreateBweSimulationGraph(Plot* plot);
+ void CreateSendSideBweSimulationGraph(Plot* plot);
+ void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreateTimestampGraph(Plot* plot);
diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index aa92a7f..fc9f707 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -71,6 +71,10 @@
DEFINE_bool(plot_outgoing_stream_bitrate,
true,
"Plot the bitrate used by each outgoing stream.");
+DEFINE_bool(plot_simulated_receiveside_bwe,
+ false,
+ "Run the receive-side bandwidth estimator with the incoming rtp "
+ "packets and plot the resulting estimate.");
DEFINE_bool(plot_simulated_sendside_bwe,
false,
"Run the send-side bandwidth estimator with the outgoing rtp and "
@@ -227,8 +231,11 @@
analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
}
+ if (FLAG_plot_simulated_receiveside_bwe) {
+ analyzer.CreateReceiveSideBweSimulationGraph(collection->AppendNewPlot());
+ }
if (FLAG_plot_simulated_sendside_bwe) {
- analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
+ analyzer.CreateSendSideBweSimulationGraph(collection->AppendNewPlot());
}
if (FLAG_plot_network_delay_feedback) {
analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
@@ -290,6 +297,7 @@
FLAG_plot_outgoing_bitrate = setting;
FLAG_plot_incoming_stream_bitrate = setting;
FLAG_plot_outgoing_stream_bitrate = setting;
+ FLAG_plot_simulated_receiveside_bwe = setting;
FLAG_plot_simulated_sendside_bwe = setting;
FLAG_plot_network_delay_feedback = setting;
FLAG_plot_fraction_loss_feedback = setting;