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}
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@ -62,10 +62,7 @@ void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
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direction == kIncomingPacket ? TriageAlertType::kIncomingCaptureTimeJump
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: TriageAlertType::kOutgoingCaptureTimeJump;
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const int64_t segment_end_us =
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parsed_log.log_segments().empty()
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? std::numeric_limits<int64_t>::max()
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: parsed_log.log_segments().front().stop_time_us();
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const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
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// Check for gaps in sequence numbers and capture timestamps.
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for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) {
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@ -133,10 +130,7 @@ void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
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? TriageAlertType::kIncomingRtcpGap
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: TriageAlertType::kOutgoingRtcpGap;
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const int64_t segment_end_us =
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parsed_log.log_segments().empty()
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? std::numeric_limits<int64_t>::max()
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: parsed_log.log_segments().front().stop_time_us();
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const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
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// TODO(terelius): The parser could provide a list of all packets, ordered
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// by time, for each direction.
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@ -202,10 +196,7 @@ void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) {
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AnalyzeTransmissionGaps(parsed_log, kIncomingPacket);
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AnalyzeTransmissionGaps(parsed_log, kOutgoingPacket);
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const int64_t segment_end_us =
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parsed_log.log_segments().empty()
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? std::numeric_limits<int64_t>::max()
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: parsed_log.log_segments().front().stop_time_us();
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const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us();
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int64_t first_occurrence = parsed_log.last_timestamp();
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constexpr double kMaxLossFraction = 0.05;
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@ -465,15 +465,21 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
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config_.begin_time_ = config_.end_time_ = 0;
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}
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RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
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<< " (LOG_START, LOG_END) segments in log.";
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RTC_LOG(LS_INFO) << "Log is "
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<< (parsed_log_.last_timestamp() -
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parsed_log_.first_timestamp()) /
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1000000
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<< " seconds long.";
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}
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EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
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const AnalyzerConfig& config)
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: parsed_log_(log), config_(config) {
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RTC_LOG(LS_INFO) << "Found " << parsed_log_.log_segments().size()
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<< " (LOG_START, LOG_END) segments in log.";
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RTC_LOG(LS_INFO) << "Log is "
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<< (parsed_log_.last_timestamp() -
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parsed_log_.first_timestamp()) /
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1000000
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<< " seconds long.";
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}
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class BitrateObserver : public RemoteBitrateObserver {
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@ -852,10 +858,7 @@ void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
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<< packets.size() << " packets in the stream.";
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continue;
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}
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int64_t segment_end_us =
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parsed_log_.log_segments().empty()
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? std::numeric_limits<int64_t>::max()
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: parsed_log_.log_segments().front().stop_time_us();
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int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
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absl::optional<uint32_t> estimated_frequency =
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EstimateRtpClockFrequency(packets, segment_end_us);
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if (!estimated_frequency)
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@ -1586,10 +1589,7 @@ void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
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"pacer delay with less than 2 packets in the stream";
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continue;
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}
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int64_t segment_end_us =
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parsed_log_.log_segments().empty()
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? std::numeric_limits<int64_t>::max()
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: parsed_log_.log_segments().front().stop_time_us();
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int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
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absl::optional<uint32_t> estimated_frequency =
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EstimateRtpClockFrequency(packets, segment_end_us);
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if (!estimated_frequency)
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@ -2050,11 +2050,7 @@ EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
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output_events_it = parsed_log_.audio_playout_events().cbegin();
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}
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absl::optional<int64_t> end_time_ms =
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parsed_log_.log_segments().empty()
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? absl::nullopt
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: absl::optional<int64_t>(
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parsed_log_.log_segments().front().stop_time_ms());
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int64_t end_time_ms = parsed_log_.first_log_segment().stop_time_ms();
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neteq_stats[ssrc] = CreateNetEqTestAndRun(
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audio_packets, &output_events_it->second, end_time_ms,
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