Revert "Create new API for RtcEventLogParser."
This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa.
Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.
Original change's description:
> Create new API for RtcEventLogParser.
>
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
>
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
>
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
> all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
> iterating over transport feedbacks and not over all RTCP packets.
> This timing changes are not visible in the plots.
>
>
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
>
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}
TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org
Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index 3dce290..2e7a79e 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -10,7 +10,7 @@
#include <iostream>
-#include "logging/rtc_event_log/rtc_event_log_parser2.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "rtc_base/flags.h"
#include "rtc_tools/event_log_visualizer/analyzer.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
@@ -143,15 +143,10 @@
false,
"Show the state ALR state on the total bitrate graph");
-DEFINE_bool(parse_unconfigured_header_extensions,
- true,
- "Attempt to parse unconfigured header extensions using the default "
- "WebRTC mapping. This can give very misleading results if the "
- "application negotiates a different mapping.");
-
-DEFINE_bool(print_triage_alerts,
- false,
- "Print triage alerts, i.e. a list of potential problems.");
+DEFINE_bool(
+ print_triage_notifications,
+ false,
+ "Print triage notifications, i.e. a list of suspicious looking events.");
void SetAllPlotFlags(bool setting);
@@ -214,13 +209,7 @@
std::string filename = argv[1];
- webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions =
- webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse;
- if (FLAG_parse_unconfigured_header_extensions) {
- header_extensions = webrtc::ParsedRtcEventLog::
- UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig;
- }
- webrtc::ParsedRtcEventLog parsed_log(header_extensions);
+ webrtc::ParsedRtcEventLog parsed_log;
if (!parsed_log.ParseFile(filename)) {
std::cerr << "Could not parse the entire log file." << std::endl;
@@ -229,34 +218,31 @@
<< std::endl;
}
- webrtc::EventLogAnalyzer analyzer(parsed_log);
- std::unique_ptr<webrtc::PlotCollection> collection(
- new webrtc::PythonPlotCollection());
+ webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
+ std::unique_ptr<webrtc::plotting::PlotCollection> collection(
+ new webrtc::plotting::PythonPlotCollection());
if (FLAG_plot_incoming_packet_sizes) {
- analyzer.CreatePacketGraph(webrtc::kIncomingPacket,
+ analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_sizes) {
- analyzer.CreatePacketGraph(webrtc::kOutgoingPacket,
+ analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_incoming_packet_count) {
- analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket,
- collection->AppendNewPlot());
+ analyzer.CreateAccumulatedPacketsGraph(
+ webrtc::PacketDirection::kIncomingPacket, collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_count) {
- analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket,
- collection->AppendNewPlot());
+ analyzer.CreateAccumulatedPacketsGraph(
+ webrtc::PacketDirection::kOutgoingPacket, collection->AppendNewPlot());
}
if (FLAG_plot_audio_playout) {
analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_level) {
- analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket,
- collection->AppendNewPlot());
- analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket,
- collection->AppendNewPlot());
+ analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_sequence_number_delta) {
analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
@@ -271,19 +257,23 @@
analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_bitrate) {
- analyzer.CreateTotalIncomingBitrateGraph(collection->AppendNewPlot());
+ analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
+ collection->AppendNewPlot(),
+ FLAG_show_detector_state,
+ FLAG_show_alr_state);
}
if (FLAG_plot_outgoing_bitrate) {
- analyzer.CreateTotalOutgoingBitrateGraph(collection->AppendNewPlot(),
- FLAG_show_detector_state,
- FLAG_show_alr_state);
+ analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
+ collection->AppendNewPlot(),
+ FLAG_show_detector_state,
+ FLAG_show_alr_state);
}
if (FLAG_plot_incoming_stream_bitrate) {
- analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket,
+ analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_stream_bitrate) {
- analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket,
+ analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_simulated_receiveside_bwe) {
@@ -299,10 +289,7 @@
analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_timestamps) {
- analyzer.CreateTimestampGraph(webrtc::kIncomingPacket,
- collection->AppendNewPlot());
- analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket,
- collection->AppendNewPlot());
+ analyzer.CreateTimestampGraph(collection->AppendNewPlot());
}
if (FLAG_plot_pacer_delay) {
analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
@@ -346,7 +333,7 @@
collection->Draw();
- if (FLAG_print_triage_alerts) {
+ if (FLAG_print_triage_notifications) {
analyzer.CreateTriageNotifications();
analyzer.PrintNotifications(stderr);
}