Add ana config to event log visualiser
BUG=webrtc:7160
Review-Url: https://codereview.webrtc.org/2695613005
Cr-Commit-Position: refs/heads/master@{#16776}
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 5c0433a..0125914 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -451,6 +451,10 @@
break;
}
case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
+ AudioNetworkAdaptationEvent ana_event;
+ ana_event.timestamp = parsed_log_.GetTimestamp(i);
+ parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
+ audio_network_adaptation_events_.push_back(ana_event);
break;
}
case ParsedRtcEventLog::UNKNOWN_EVENT: {
@@ -532,6 +536,21 @@
return name.str();
}
+void EventLogAnalyzer::FillAudioEncoderTimeSeries(
+ Plot* plot,
+ rtc::FunctionView<rtc::Optional<float>(
+ const AudioNetworkAdaptationEvent& ana_event)> get_y) const {
+ plot->series_list_.push_back(TimeSeries());
+ plot->series_list_.back().style = LINE_DOT_GRAPH;
+ for (auto& ana_event : audio_network_adaptation_events_) {
+ rtc::Optional<float> y = get_y(ana_event);
+ if (y) {
+ float x = static_cast<float>(ana_event.timestamp - begin_time_) / 1000000;
+ plot->series_list_.back().points.emplace_back(x, *y);
+ }
+ }
+}
+
void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
Plot* plot) {
for (auto& kv : rtp_packets_) {
@@ -1275,5 +1294,92 @@
plot->SetSuggestedYAxis(0, 1, "Timestamp (90khz)", kBottomMargin, kTopMargin);
plot->SetTitle("Timestamps");
}
+
+void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.bitrate_bps)
+ return rtc::Optional<float>(
+ static_cast<float>(*ana_event.config.bitrate_bps));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder target bitrate";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
+ plot->SetTitle("Reported audio encoder target bitrate");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.frame_length_ms)
+ return rtc::Optional<float>(
+ static_cast<float>(*ana_event.config.frame_length_ms));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder frame length";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
+ plot->SetTitle("Reported audio encoder frame length");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderUplinkPacketLossFractionGraph(
+ Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.uplink_packet_loss_fraction)
+ return rtc::Optional<float>(static_cast<float>(
+ *ana_event.config.uplink_packet_loss_fraction));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder uplink packet loss fraction";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
+ kTopMargin);
+ plot->SetTitle("Reported audio encoder lost packets");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.enable_fec)
+ return rtc::Optional<float>(
+ static_cast<float>(*ana_event.config.enable_fec));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder FEC";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
+ plot->SetTitle("Reported audio encoder FEC");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.enable_dtx)
+ return rtc::Optional<float>(
+ static_cast<float>(*ana_event.config.enable_dtx));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder DTX";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
+ plot->SetTitle("Reported audio encoder DTX");
+}
+
+void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
+ FillAudioEncoderTimeSeries(
+ plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
+ if (ana_event.config.num_channels)
+ return rtc::Optional<float>(
+ static_cast<float>(*ana_event.config.num_channels));
+ return rtc::Optional<float>();
+ });
+ plot->series_list_.back().label = "Audio encoder number of channels";
+ plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
+ plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
+ kBottomMargin, kTopMargin);
+ plot->SetTitle("Reported audio encoder number of channels");
+}
} // namespace plotting
} // namespace webrtc
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h
index f0557a2..c15cb75 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.h
+++ b/webrtc/tools/event_log_visualizer/analyzer.h
@@ -18,6 +18,7 @@
#include <utility>
#include <vector>
+#include "webrtc/base/function_view.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
@@ -52,6 +53,11 @@
int32_t expected_packets;
};
+struct AudioNetworkAdaptationEvent {
+ uint64_t timestamp;
+ AudioNetworkAdaptor::EncoderRuntimeConfig config;
+};
+
class EventLogAnalyzer {
public:
// The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
@@ -87,6 +93,13 @@
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreateTimestampGraph(Plot* plot);
+ void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
+ void CreateAudioEncoderFrameLengthGraph(Plot* plot);
+ void CreateAudioEncoderUplinkPacketLossFractionGraph(Plot* plot);
+ void CreateAudioEncoderEnableFecGraph(Plot* plot);
+ void CreateAudioEncoderEnableDtxGraph(Plot* plot);
+ void CreateAudioEncoderNumChannelsGraph(Plot* plot);
+
// Returns a vector of capture and arrival timestamps for the video frames
// of the stream with the most number of frames.
std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
@@ -127,6 +140,11 @@
std::string GetStreamName(StreamId) const;
+ void FillAudioEncoderTimeSeries(
+ Plot* plot,
+ rtc::FunctionView<rtc::Optional<float>(
+ const AudioNetworkAdaptationEvent& ana_event)> get_y) const;
+
const ParsedRtcEventLog& parsed_log_;
// A list of SSRCs we are interested in analysing.
@@ -152,6 +170,8 @@
// A list of all updates from the send-side loss-based bandwidth estimator.
std::vector<LossBasedBweUpdate> bwe_loss_updates_;
+ std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
+
// Window and step size used for calculating moving averages, e.g. bitrate.
// The generated data points will be |step_| microseconds apart.
// Only events occuring at most |window_duration_| microseconds before the
diff --git a/webrtc/tools/event_log_visualizer/main.cc b/webrtc/tools/event_log_visualizer/main.cc
index 13a5b8a..2f5ecd6 100644
--- a/webrtc/tools/event_log_visualizer/main.cc
+++ b/webrtc/tools/event_log_visualizer/main.cc
@@ -62,6 +62,21 @@
DEFINE_bool(plot_timestamps,
false,
"Plot the rtp timestamps of all rtp and rtcp packets over time.");
+DEFINE_bool(audio_encoder_bitrate_bps,
+ false,
+ "Plot the audio encoder target bitrate.");
+DEFINE_bool(audio_encoder_frame_length_ms,
+ false,
+ "Plot the audio encoder frame length.");
+DEFINE_bool(
+ audio_encoder_uplink_packet_loss_fraction,
+ false,
+ "Plot the uplink packet loss fraction which is send to the audio encoder.");
+DEFINE_bool(audio_encoder_fec, false, "Plot the audio encoder FEC.");
+DEFINE_bool(audio_encoder_dtx, false, "Plot the audio encoder DTX.");
+DEFINE_bool(audio_encoder_num_channels,
+ false,
+ "Plot the audio encoder number of channels.");
DEFINE_string(
force_fieldtrials,
"",
@@ -187,6 +202,31 @@
analyzer.CreateTimestampGraph(collection->AppendNewPlot());
}
+ if (FLAGS_plot_all || FLAGS_audio_encoder_bitrate_bps) {
+ analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
+ }
+
+ if (FLAGS_plot_all || FLAGS_audio_encoder_frame_length_ms) {
+ analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot());
+ }
+
+ if (FLAGS_plot_all || FLAGS_audio_encoder_uplink_packet_loss_fraction) {
+ analyzer.CreateAudioEncoderUplinkPacketLossFractionGraph(
+ collection->AppendNewPlot());
+ }
+
+ if (FLAGS_plot_all || FLAGS_audio_encoder_fec) {
+ analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
+ }
+
+ if (FLAGS_plot_all || FLAGS_audio_encoder_dtx) {
+ analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
+ }
+
+ if (FLAGS_plot_all || FLAGS_audio_encoder_num_channels) {
+ analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
+ }
+
collection->Draw();
return 0;