Move webrtc/{tools => rtc_tools}

webrtc/tools/event_log_visualizer/main.cc

Index: webrtc/tools/event_log_visualizer/main.cc
diff --git a/webrtc/tools/event_log_visualizer/main.cc b/webrtc/tools/event_log_visualizer/main.cc
deleted file mode 100644
index 91d599f2b31e481798f922e4a8c1dce416cd2063..0000000000000000000000000000000000000000
--- a/webrtc/tools/event_log_visualizer/main.cc
+++ /dev/null
@@ -1,250 +0,0 @@
-/*
- *  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/base/flags.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/tools/event_log_visualizer/analyzer.h"
-#include "webrtc/tools/event_log_visualizer/plot_base.h"
-#include "webrtc/tools/event_log_visualizer/plot_python.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,
-            false,
-            "Plot bar graph showing the size of each packet.");
-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.");
-DEFINE_bool(
-    plot_delay_change,
-    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_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.");
-DEFINE_bool(plot_network_delay_feedback,
-            false,
-            "Compute network delay based on sent packets and the received "
-            "transport feedback.");
-DEFINE_bool(plot_fraction_loss,
-            false,
-            "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,
-            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_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");
-
-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";
-  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
-  if (FLAG_help) {
-    rtc::FlagList::Print(nullptr, false);
-    return 0;
-  }
-
-  if (argc != 2) {
-    // Print usage information.
-    std::cout << usage;
-    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_all || FLAG_plot_audio_playout) {
-    analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || 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_all || FLAG_plot_delay_change) {
-    if (FLAG_incoming) {
-      analyzer.CreateDelayChangeGraph(collection->AppendNewPlot());
-    }
-  }
-
-  if (FLAG_plot_all || FLAG_plot_accumulated_delay_change) {
-    if (FLAG_incoming) {
-      analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot());
-    }
-  }
-
-  if (FLAG_plot_all || FLAG_plot_fraction_loss) {
-    analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
-    analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_plot_total_bitrate) {
-    if (FLAG_incoming) {
-      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
-                                       collection->AppendNewPlot());
-    }
-    if (FLAG_outgoing) {
-      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
-                                       collection->AppendNewPlot());
-    }
-  }
-
-  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) {
-    analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_plot_network_delay_feedback) {
-    analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_plot_timestamps) {
-    analyzer.CreateTimestampGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_audio_encoder_bitrate_bps) {
-    analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_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_all || FLAG_audio_encoder_fec) {
-    analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_audio_encoder_dtx) {
-    analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || FLAG_audio_encoder_num_channels) {
-    analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
-  }
-
-  if (FLAG_plot_all || 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;
-}