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Revert of Visualization tool for WebrtcEventLogs (patchset #9 id:160001 of https://codereview.webrtc.org/1995523002/ )

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Reason for revert:
Reverting while investigating a downstream build failure.

Original issue's description:
> Initial version of the local visualization tool for WebrtcEventLogs.
>
> Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.
>
> Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.
>
> Committed: https://crrev.com/a478786ef1513790194792010f766324a469db4d
> Cr-Commit-Position: refs/heads/master@{#13443}

TBR=stefan@webrtc.org,aleloi@webrtc.org,phoglund@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2147453002
Cr-Commit-Position: refs/heads/master@{#13445}
This commit is contained in:
terelius
2016-07-12 05:11:21 -07:00
committed by Commit bot
parent d658edeeed
commit a44f28da45
10 changed files with 1 additions and 1181 deletions
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138
webrtc/tools/event_log_visualizer/generate_timeseries.cc
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@ -1,138 +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 "gflags/gflags.h"
#include "webrtc/call/rtc_event_log_parser.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_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.");
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";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2) {
// Print usage information.
std::cout << google::ProgramUsage();
return 0;
}
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 (FLAGS_plot_all || FLAGS_plot_packets) {
if (FLAGS_incoming) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
collection->append_new_plot());
}
if (FLAGS_outgoing) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->append_new_plot());
}
}
if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
analyzer.CreatePlayoutGraph(collection->append_new_plot());
}
if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
if (FLAGS_incoming) {
analyzer.CreateSequenceNumberGraph(collection->append_new_plot());
}
}
if (FLAGS_plot_all || FLAGS_plot_delay_change) {
if (FLAGS_incoming) {
analyzer.CreateDelayChangeGraph(collection->append_new_plot());
}
}
if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
if (FLAGS_incoming) {
analyzer.CreateAccumulatedDelayChangeGraph(collection->append_new_plot());
}
}
if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
if (FLAGS_incoming) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
collection->append_new_plot());
}
if (FLAGS_outgoing) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->append_new_plot());
}
}
if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
if (FLAGS_incoming) {
analyzer.CreateStreamBitrateGraph(
webrtc::PacketDirection::kIncomingPacket,
collection->append_new_plot());
}
if (FLAGS_outgoing) {
analyzer.CreateStreamBitrateGraph(
webrtc::PacketDirection::kOutgoingPacket,
collection->append_new_plot());
}
}
collection->draw();
return 0;
}