Index: webrtc/rtc_tools/event_log_visualizer/main.cc |
diff --git a/webrtc/rtc_tools/event_log_visualizer/main.cc b/webrtc/rtc_tools/event_log_visualizer/main.cc |
index 7df0374484aec628b85e76f2884ab14054c189e7..8db1f9753202346cc8d03183f1495cd979268b14 100644 |
--- a/webrtc/rtc_tools/event_log_visualizer/main.cc |
+++ b/webrtc/rtc_tools/event_log_visualizer/main.cc |
@@ -18,64 +18,75 @@ |
#include "webrtc/test/field_trial.h" |
#include "webrtc/test/testsupport/fileutils.h" |
+DEFINE_string(plot_profile, |
+ "default", |
+ "A profile that selects a certain subset of the plots. Currently " |
+ "defined profiles are \"all\", \"none\" and \"default\""); |
+ |
DEFINE_bool(incoming, true, "Plot statistics for incoming packets."); |
DEFINE_bool(outgoing, true, "Plot statistics for outgoing packets."); |
-DEFINE_bool(plot_all, true, "Plot all different data types."); |
-DEFINE_bool(plot_packets, |
+DEFINE_bool(plot_packet_sizes, |
false, |
"Plot bar graph showing the size of each packet."); |
+DEFINE_bool(plot_packet_count, |
+ false, |
+ "Plot the accumulated number of packets for each stream."); |
DEFINE_bool(plot_audio_playout, |
false, |
"Plot bar graph showing the time between each audio playout."); |
DEFINE_bool(plot_audio_level, |
false, |
- "Plot line graph showing the audio level."); |
-DEFINE_bool( |
- plot_sequence_number, |
- false, |
- "Plot the difference in sequence number between consecutive packets."); |
+ "Plot line graph showing the audio level of incoming audio."); |
+DEFINE_bool(plot_sequence_number, |
+ false, |
+ "Plot the sequence number difference between consecutive incoming " |
+ "packets."); |
DEFINE_bool( |
- plot_delay_change, |
+ plot_incoming_delay_delta, |
false, |
"Plot the difference in 1-way path delay between consecutive packets."); |
-DEFINE_bool(plot_accumulated_delay_change, |
- false, |
- "Plot the accumulated 1-way path delay change, or the path delay " |
- "change compared to the first packet."); |
+DEFINE_bool(plot_incoming_delay, |
+ true, |
+ "Plot the 1-way path delay for incoming packets, normalized so " |
+ "that the first packet has delay 0."); |
+DEFINE_bool(plot_incoming_loss_rate, |
+ true, |
+ "Compute the loss rate for incoming packets using a method that's " |
+ "similar to the one used for RTCP SR and RR fraction lost. Note " |
+ "that the loss rate can be negative if packets are duplicated or " |
+ "reordered."); |
DEFINE_bool(plot_total_bitrate, |
- false, |
+ true, |
"Plot the total bitrate used by all streams."); |
-DEFINE_bool(plot_stream_bitrate, |
- false, |
- "Plot the bitrate used by each stream."); |
-DEFINE_bool(plot_bwe, |
+DEFINE_bool(plot_stream_bitrate, true, "Plot the bitrate used by each stream."); |
+DEFINE_bool(plot_simulated_sendside_bwe, |
false, |
- "Run the bandwidth estimator with the logged rtp and rtcp and plot " |
- "the output."); |
+ "Run the send-side bandwidth estimator with the outgoing rtp and " |
+ "incoming rtcp and plot the resulting estimate."); |
DEFINE_bool(plot_network_delay_feedback, |
- false, |
+ true, |
"Compute network delay based on sent packets and the received " |
"transport feedback."); |
-DEFINE_bool(plot_fraction_loss, |
- false, |
+DEFINE_bool(plot_fraction_loss_feedback, |
+ true, |
"Plot packet loss in percent for outgoing packets (as perceived by " |
"the send-side bandwidth estimator)."); |
DEFINE_bool(plot_timestamps, |
false, |
"Plot the rtp timestamps of all rtp and rtcp packets over time."); |
-DEFINE_bool(audio_encoder_bitrate_bps, |
+DEFINE_bool(plot_audio_encoder_bitrate_bps, |
false, |
"Plot the audio encoder target bitrate."); |
-DEFINE_bool(audio_encoder_frame_length_ms, |
+DEFINE_bool(plot_audio_encoder_frame_length_ms, |
false, |
"Plot the audio encoder frame length."); |
DEFINE_bool( |
- audio_encoder_uplink_packet_loss_fraction, |
+ plot_audio_encoder_packet_loss, |
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, |
+ "Plot the uplink packet loss fraction which is sent to the audio encoder."); |
+DEFINE_bool(plot_audio_encoder_fec, false, "Plot the audio encoder FEC."); |
+DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX."); |
+DEFINE_bool(plot_audio_encoder_num_channels, |
false, |
"Plot the audio encoder number of channels."); |
DEFINE_bool(plot_audio_jitter_buffer, |
@@ -90,10 +101,13 @@ DEFINE_string( |
"trials are separated by \"/\""); |
DEFINE_bool(help, false, "prints this message"); |
-DEFINE_bool( |
- show_detector_state, |
- false, |
- "Mark the delay based bwe detector state on the total bitrate graph"); |
+DEFINE_bool(show_detector_state, |
+ false, |
+ "Show the state of the delay based BWE detector on the total " |
+ "bitrate graph"); |
+ |
+void SetAllPlotFlags(bool setting); |
+ |
int main(int argc, char* argv[]) { |
std::string program_name = argv[0]; |
@@ -102,7 +116,24 @@ int main(int argc, char* argv[]) { |
"Example usage:\n" + |
program_name + " <logfile> | python\n" + "Run " + program_name + |
" --help for a list of command line options\n"; |
+ |
+ // Parse command line flags without removing them. We're only interested in |
+ // the |plot_profile| flag. |
+ rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, false); |
+ if (strcmp(FLAG_plot_profile, "all") == 0) { |
+ SetAllPlotFlags(true); |
+ } else if (strcmp(FLAG_plot_profile, "none") == 0) { |
+ SetAllPlotFlags(false); |
+ } else if (strcmp(FLAG_plot_profile, "default") == 0) { |
+ // Do nothing. |
+ } else { |
+ rtc::Flag* plot_profile_flag = rtc::FlagList::Lookup("plot_profile"); |
+ RTC_CHECK(plot_profile_flag); |
+ plot_profile_flag->Print(false); |
+ } |
+ // Parse the remaining flags. They are applied relative to the chosen profile. |
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true); |
+ |
if (argc != 2 || FLAG_help) { |
// Print usage information. |
std::cout << usage; |
@@ -129,55 +160,47 @@ int main(int argc, char* argv[]) { |
std::unique_ptr<webrtc::plotting::PlotCollection> collection( |
new webrtc::plotting::PythonPlotCollection()); |
- if (FLAG_plot_all || FLAG_plot_packets) { |
+ if (FLAG_plot_packet_sizes) { |
if (FLAG_incoming) { |
analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket, |
collection->AppendNewPlot()); |
+ } |
+ if (FLAG_outgoing) { |
+ analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket, |
+ collection->AppendNewPlot()); |
+ } |
+ } |
+ if (FLAG_plot_packet_count) { |
+ if (FLAG_incoming) { |
minyue-webrtc
2017/07/21 15:08:43
I would suggest remove FLAG_incoming and FLAG_outg
terelius
2017/08/08 12:15:59
Done.
|
analyzer.CreateAccumulatedPacketsGraph( |
webrtc::PacketDirection::kIncomingPacket, |
collection->AppendNewPlot()); |
} |
if (FLAG_outgoing) { |
- analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket, |
- collection->AppendNewPlot()); |
analyzer.CreateAccumulatedPacketsGraph( |
webrtc::PacketDirection::kOutgoingPacket, |
collection->AppendNewPlot()); |
} |
} |
- |
- if (FLAG_plot_all || FLAG_plot_audio_playout) { |
+ if (FLAG_plot_audio_playout) { |
analyzer.CreatePlayoutGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_audio_level) { |
+ if (FLAG_plot_audio_level) { |
analyzer.CreateAudioLevelGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_sequence_number) { |
- if (FLAG_incoming) { |
- analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot()); |
- } |
+ if (FLAG_plot_sequence_number && FLAG_incoming) { |
+ analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_delay_change) { |
- if (FLAG_incoming) { |
- analyzer.CreateDelayChangeGraph(collection->AppendNewPlot()); |
- } |
+ if (FLAG_plot_incoming_delay_delta && FLAG_incoming) { |
+ analyzer.CreateIncomingDelayDeltaGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_accumulated_delay_change) { |
- if (FLAG_incoming) { |
- analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot()); |
- } |
+ if (FLAG_plot_incoming_delay && FLAG_incoming) { |
+ analyzer.CreateIncomingDelayGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_fraction_loss) { |
- analyzer.CreateFractionLossGraph(collection->AppendNewPlot()); |
+ if (FLAG_plot_incoming_loss_rate && FLAG_incoming) { |
analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_total_bitrate) { |
+ if (FLAG_plot_total_bitrate) { |
if (FLAG_incoming) { |
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket, |
collection->AppendNewPlot(), |
@@ -189,8 +212,7 @@ int main(int argc, char* argv[]) { |
FLAG_show_detector_state); |
} |
} |
- |
- if (FLAG_plot_all || FLAG_plot_stream_bitrate) { |
+ if (FLAG_plot_stream_bitrate) { |
if (FLAG_incoming) { |
analyzer.CreateStreamBitrateGraph( |
webrtc::PacketDirection::kIncomingPacket, |
@@ -202,45 +224,37 @@ int main(int argc, char* argv[]) { |
collection->AppendNewPlot()); |
} |
} |
- |
- if (FLAG_plot_all || FLAG_plot_bwe) { |
+ if (FLAG_plot_simulated_sendside_bwe) { |
analyzer.CreateBweSimulationGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_network_delay_feedback) { |
+ if (FLAG_plot_network_delay_feedback) { |
analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_timestamps) { |
+ if (FLAG_plot_fraction_loss_feedback && FLAG_outgoing) { |
+ analyzer.CreateFractionLossGraph(collection->AppendNewPlot()); |
+ } |
+ if (FLAG_plot_timestamps) { |
analyzer.CreateTimestampGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_bitrate_bps) { |
+ if (FLAG_plot_audio_encoder_bitrate_bps) { |
analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_frame_length_ms) { |
+ if (FLAG_plot_audio_encoder_frame_length_ms) { |
analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_uplink_packet_loss_fraction) { |
- analyzer.CreateAudioEncoderUplinkPacketLossFractionGraph( |
- collection->AppendNewPlot()); |
+ if (FLAG_plot_audio_encoder_packet_loss) { |
+ analyzer.CreateAudioEncoderPacketLossGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_fec) { |
+ if (FLAG_plot_audio_encoder_fec) { |
analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_dtx) { |
+ if (FLAG_plot_audio_encoder_dtx) { |
analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_audio_encoder_num_channels) { |
+ if (FLAG_plot_audio_encoder_num_channels) { |
analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot()); |
} |
- |
- if (FLAG_plot_all || FLAG_plot_audio_jitter_buffer) { |
+ if (FLAG_plot_audio_jitter_buffer) { |
analyzer.CreateAudioJitterBufferGraph( |
webrtc::test::ResourcePath( |
"audio_processing/conversational_speech/EN_script2_F_sp2_B1", |
@@ -252,3 +266,28 @@ int main(int argc, char* argv[]) { |
return 0; |
} |
+ |
+ |
+void SetAllPlotFlags(bool setting) { |
+ FLAG_plot_packet_sizes = setting; |
+ FLAG_plot_packet_count = setting; |
+ FLAG_plot_audio_playout = setting; |
+ FLAG_plot_audio_level = setting; |
+ FLAG_plot_sequence_number = setting; |
+ FLAG_plot_incoming_delay_delta = setting; |
+ FLAG_plot_incoming_delay = setting; |
+ FLAG_plot_incoming_loss_rate = setting; |
+ FLAG_plot_total_bitrate = setting; |
+ FLAG_plot_stream_bitrate = setting; |
+ FLAG_plot_simulated_sendside_bwe = setting; |
+ FLAG_plot_network_delay_feedback = setting; |
+ FLAG_plot_fraction_loss_feedback = setting; |
+ FLAG_plot_timestamps = setting; |
+ FLAG_plot_audio_encoder_bitrate_bps = setting; |
+ FLAG_plot_audio_encoder_frame_length_ms = setting; |
+ FLAG_plot_audio_encoder_packet_loss = setting; |
+ FLAG_plot_audio_encoder_fec = setting; |
+ FLAG_plot_audio_encoder_dtx = setting; |
+ FLAG_plot_audio_encoder_num_channels = setting; |
+ FLAG_plot_audio_jitter_buffer = setting; |
+} |