Replace inconsistent log_segments() function in RTC event log parser
Bug: webrtc:11566
Change-Id: I739bbc29ae5423f3fedcc08e991e27fa0af840c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176081
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31371}
diff --git a/rtc_tools/rtc_event_log_visualizer/alerts.cc b/rtc_tools/rtc_event_log_visualizer/alerts.cc
index f2d9564..86372de 100644
--- a/rtc_tools/rtc_event_log_visualizer/alerts.cc
+++ b/rtc_tools/rtc_event_log_visualizer/alerts.cc
@@ -62,10 +62,7 @@
direction == kIncomingPacket ? TriageAlertType::kIncomingCaptureTimeJump
: TriageAlertType::kOutgoingCaptureTimeJump;
- const int64_t segment_end_us =
- parsed_log.log_segments().empty()
- ? std::numeric_limits<int64_t>::max()
- : parsed_log.log_segments().front().stop_time_us();
+ const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
// Check for gaps in sequence numbers and capture timestamps.
for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
@@ -133,10 +130,7 @@
? TriageAlertType::kIncomingRtcpGap
: TriageAlertType::kOutgoingRtcpGap;
- const int64_t segment_end_us =
- parsed_log.log_segments().empty()
- ? std::numeric_limits<int64_t>::max()
- : parsed_log.log_segments().front().stop_time_us();
+ const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
// TODO(terelius): The parser could provide a list of all packets, ordered
// by time, for each direction.
@@ -202,10 +196,7 @@
AnalyzeTransmissionGaps(parsed_log, kIncomingPacket);
AnalyzeTransmissionGaps(parsed_log, kOutgoingPacket);
- const int64_t segment_end_us =
- parsed_log.log_segments().empty()
- ? std::numeric_limits<int64_t>::max()
- : parsed_log.log_segments().front().stop_time_us();
+ const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
int64_t first_occurrence = parsed_log.last_timestamp();
constexpr double kMaxLossFraction = 0.05;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 9a9a455..a0e441c 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -465,15 +465,21 @@
config_.begin_time_ = config_.end_time_ = 0;
}
- RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
- << " (LOG_START, LOG_END) segments in log.";
+ RTC_LOG(LS_INFO) << "Log is "
+ << (parsed_log_.last_timestamp() -
+ parsed_log_.first_timestamp()) /
+ 1000000
+ << " seconds long.";
}
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
const AnalyzerConfig& config)
: parsed_log_(log), config_(config) {
- RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
- << " (LOG_START, LOG_END) segments in log.";
+ RTC_LOG(LS_INFO) << "Log is "
+ << (parsed_log_.last_timestamp() -
+ parsed_log_.first_timestamp()) /
+ 1000000
+ << " seconds long.";
}
class BitrateObserver : public RemoteBitrateObserver {
@@ -852,10 +858,7 @@
<< packets.size() << " packets in the stream.";
continue;
}
- int64_t segment_end_us =
- parsed_log_.log_segments().empty()
- ? std::numeric_limits<int64_t>::max()
- : parsed_log_.log_segments().front().stop_time_us();
+ int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, segment_end_us);
if (!estimated_frequency)
@@ -1586,10 +1589,7 @@
"pacer delay with less than 2 packets in the stream";
continue;
}
- int64_t segment_end_us =
- parsed_log_.log_segments().empty()
- ? std::numeric_limits<int64_t>::max()
- : parsed_log_.log_segments().front().stop_time_us();
+ int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, segment_end_us);
if (!estimated_frequency)
@@ -2050,11 +2050,7 @@
output_events_it = parsed_log_.audio_playout_events().cbegin();
}
- absl::optional<int64_t> end_time_ms =
- parsed_log_.log_segments().empty()
- ? absl::nullopt
- : absl::optional<int64_t>(
- parsed_log_.log_segments().front().stop_time_ms());
+ int64_t end_time_ms = parsed_log_.first_log_segment().stop_time_ms();
neteq_stats[ssrc] = CreateNetEqTestAndRun(
audio_packets, &output_events_it->second, end_time_ms,