Improved UI for event_log_analyzer tool

webrtc/rtc_tools/event_log_visualizer/main.cc

 /*
  *  Copyright (c) 2016 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 <iostream>
 
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_tools/event_log_visualizer/analyzer.h"
 #include "webrtc/rtc_tools/event_log_visualizer/plot_base.h"
 #include "webrtc/rtc_tools/event_log_visualizer/plot_python.h"
 #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 packet.");
+            "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 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.");
+            "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_sequence_number,
-    false,
-    "Plot the difference in sequence number between consecutive 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,
             false,
             "Plot the audio jitter buffer delay profile.");
 DEFINE_string(
     force_fieldtrials,
     "",
     "Field trials control experimental feature code which can be forced. "
     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
     " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
     "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];
   std::string usage =
       "A tool for visualizing WebRTC event logs.\n"
       "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;
     if (FLAG_help)
       rtc::FlagList::Print(nullptr, false);
     return 0;
   }
 
   webrtc::test::SetExecutablePath(argv[0]);
   webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
 
   std::string filename = argv[1];
 
   webrtc::ParsedRtcEventLog parsed_log;
 
   if (!parsed_log.ParseFile(filename)) {
     std::cerr << "Could not parse the entire log file." << std::endl;
     std::cerr << "Proceeding to analyze the first "
               << parsed_log.GetNumberOfEvents() << " events in the file."
               << std::endl;
   }
 
   webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
   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_incoming_packet_sizes) {
+    analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
+                               collection->AppendNewPlot());
   }
-
-  if (FLAG_plot_all || FLAG_plot_audio_playout) {
+  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_incoming_bitrate) {
+    analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
+                                     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_outgoing_bitrate) {
+    analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
+                                     collection->AppendNewPlot(),
+                                     FLAG_show_detector_state);
   }
-
-  if (FLAG_plot_all || FLAG_plot_bwe) {
+  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",
             "wav"),
         48000, collection->AppendNewPlot());
   }
 
   collection->Draw();
 
   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;
+}