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}
This commit is contained in:
Björn Terelius
2020-05-27 20:25:06 +02:00
committed by Commit Bot
parent 0ee4ee85dd
commit 0d1b28cf09
6 changed files with 59 additions and 59 deletions

View File

@ -62,10 +62,7 @@ void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
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 @@ void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
? 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 @@ void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) {
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;

View File

@ -465,15 +465,21 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
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 @@ void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
<< 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 @@ void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
"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 @@ EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
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,