| 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..12b55e6d24481bfde83d9dc32a2b12e8e210e416 100644
|
| --- a/webrtc/rtc_tools/event_log_visualizer/main.cc
|
| +++ b/webrtc/rtc_tools/event_log_visualizer/main.cc
|
| @@ -18,64 +18,87 @@
|
| #include "webrtc/test/field_trial.h"
|
| #include "webrtc/test/testsupport/fileutils.h"
|
|
|
| -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_string(plot_profile,
|
| + "default",
|
| + "A profile that selects a certain subset of the plots. Currently "
|
| + "defined profiles are \"all\", \"none\" and \"default\"");
|
| +
|
| +DEFINE_bool(plot_incoming_packet_sizes,
|
| + false,
|
| + "Plot bar graph showing the size of each incoming packet.");
|
| +DEFINE_bool(plot_outgoing_packet_sizes,
|
| + false,
|
| + "Plot bar graph showing the size of each outgoing packet.");
|
| +DEFINE_bool(plot_incoming_packet_count,
|
| + false,
|
| + "Plot the accumulated number of packets for each incoming stream.");
|
| +DEFINE_bool(plot_outgoing_packet_count,
|
| false,
|
| - "Plot bar graph showing the size of each packet.");
|
| + "Plot the accumulated number of packets for each outgoing 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_incoming_sequence_number_delta,
|
| + 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,
|
| +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_incoming_bitrate,
|
| + true,
|
| + "Plot the total bitrate used by all incoming streams.");
|
| +DEFINE_bool(plot_outgoing_bitrate,
|
| + true,
|
| + "Plot the total bitrate used by all outgoing streams.");
|
| +DEFINE_bool(plot_incoming_stream_bitrate,
|
| + true,
|
| + "Plot the bitrate used by each incoming stream.");
|
| +DEFINE_bool(plot_outgoing_stream_bitrate,
|
| + true,
|
| + "Plot the bitrate used by each outgoing stream.");
|
| +DEFINE_bool(plot_simulated_sendside_bwe,
|
| false,
|
| - "Plot the accumulated 1-way path delay change, or the path delay "
|
| - "change compared to the first packet.");
|
| -DEFINE_bool(plot_total_bitrate,
|
| - false,
|
| - "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,
|
| - 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 +113,13 @@ 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 +128,24 @@ 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;
|
| @@ -129,118 +172,89 @@ 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_incoming) {
|
| - analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot());
|
| - 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_incoming_packet_sizes) {
|
| + analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
|
| + collection->AppendNewPlot());
|
| + }
|
| + if (FLAG_plot_outgoing_packet_sizes) {
|
| + analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| + collection->AppendNewPlot());
|
| + }
|
| + if (FLAG_plot_incoming_packet_count) {
|
| + analyzer.CreateAccumulatedPacketsGraph(
|
| + webrtc::PacketDirection::kIncomingPacket, collection->AppendNewPlot());
|
| + }
|
| + if (FLAG_plot_outgoing_packet_count) {
|
| + analyzer.CreateAccumulatedPacketsGraph(
|
| + webrtc::PacketDirection::kOutgoingPacket, collection->AppendNewPlot());
|
| + }
|
| + 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_incoming_sequence_number_delta) {
|
| + 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) {
|
| + 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) {
|
| + analyzer.CreateIncomingDelayGraph(collection->AppendNewPlot());
|
| }
|
| -
|
| - if (FLAG_plot_all || FLAG_plot_fraction_loss) {
|
| - analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
|
| + if (FLAG_plot_incoming_loss_rate) {
|
| analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
|
| }
|
| -
|
| - if (FLAG_plot_all || FLAG_plot_total_bitrate) {
|
| - if (FLAG_incoming) {
|
| - analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot(),
|
| - FLAG_show_detector_state);
|
| - }
|
| - if (FLAG_outgoing) {
|
| - analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| - collection->AppendNewPlot(),
|
| - FLAG_show_detector_state);
|
| - }
|
| - }
|
| -
|
| - if (FLAG_plot_all || FLAG_plot_stream_bitrate) {
|
| - if (FLAG_incoming) {
|
| - analyzer.CreateStreamBitrateGraph(
|
| - webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - if (FLAG_outgoing) {
|
| - analyzer.CreateStreamBitrateGraph(
|
| - webrtc::PacketDirection::kOutgoingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAG_plot_all || FLAG_plot_bwe) {
|
| + if (FLAG_plot_incoming_bitrate) {
|
| + analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
|
| + collection->AppendNewPlot(),
|
| + FLAG_show_detector_state);
|
| + }
|
| + if (FLAG_plot_outgoing_bitrate) {
|
| + analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| + collection->AppendNewPlot(),
|
| + FLAG_show_detector_state);
|
| + }
|
| + if (FLAG_plot_incoming_stream_bitrate) {
|
| + analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
|
| + collection->AppendNewPlot());
|
| + }
|
| + if (FLAG_plot_outgoing_stream_bitrate) {
|
| + analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| + collection->AppendNewPlot());
|
| + }
|
| + 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) {
|
| + 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 +266,32 @@ int main(int argc, char* argv[]) {
|
|
|
| return 0;
|
| }
|
| +
|
| +
|
| +void SetAllPlotFlags(bool setting) {
|
| + FLAG_plot_incoming_packet_sizes = setting;
|
| + FLAG_plot_outgoing_packet_sizes = setting;
|
| + FLAG_plot_incoming_packet_count = setting;
|
| + FLAG_plot_outgoing_packet_count = setting;
|
| + FLAG_plot_audio_playout = setting;
|
| + FLAG_plot_audio_level = setting;
|
| + FLAG_plot_incoming_sequence_number_delta = setting;
|
| + FLAG_plot_incoming_delay_delta = setting;
|
| + FLAG_plot_incoming_delay = setting;
|
| + FLAG_plot_incoming_loss_rate = setting;
|
| + FLAG_plot_incoming_bitrate = setting;
|
| + FLAG_plot_outgoing_bitrate = setting;
|
| + FLAG_plot_incoming_stream_bitrate = setting;
|
| + FLAG_plot_outgoing_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;
|
| +}
|
|
|