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Searched refs:parsed_log (Results 1 – 14 of 14) sorted by relevance

/external/webrtc/rtc_tools/rtc_event_log_visualizer/
Danalyzer_common.cc16 bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log, in IsRtxSsrc() argument
20 return parsed_log.incoming_rtx_ssrcs().find(ssrc) != in IsRtxSsrc()
21 parsed_log.incoming_rtx_ssrcs().end(); in IsRtxSsrc()
23 return parsed_log.outgoing_rtx_ssrcs().find(ssrc) != in IsRtxSsrc()
24 parsed_log.outgoing_rtx_ssrcs().end(); in IsRtxSsrc()
28 bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log, in IsVideoSsrc() argument
32 return parsed_log.incoming_video_ssrcs().find(ssrc) != in IsVideoSsrc()
33 parsed_log.incoming_video_ssrcs().end(); in IsVideoSsrc()
35 return parsed_log.outgoing_video_ssrcs().find(ssrc) != in IsVideoSsrc()
36 parsed_log.outgoing_video_ssrcs().end(); in IsVideoSsrc()
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Danalyze_audio.h27 void CreateAudioEncoderTargetBitrateGraph(const ParsedRtcEventLog& parsed_log,
30 void CreateAudioEncoderFrameLengthGraph(const ParsedRtcEventLog& parsed_log,
33 void CreateAudioEncoderPacketLossGraph(const ParsedRtcEventLog& parsed_log,
36 void CreateAudioEncoderEnableFecGraph(const ParsedRtcEventLog& parsed_log,
39 void CreateAudioEncoderEnableDtxGraph(const ParsedRtcEventLog& parsed_log,
42 void CreateAudioEncoderNumChannelsGraph(const ParsedRtcEventLog& parsed_log,
48 NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log,
53 void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log,
59 const ParsedRtcEventLog& parsed_log,
66 const ParsedRtcEventLog& parsed_log,
Dalerts.cc28 void TriageHelper::AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log, in AnalyzeStreamGaps() argument
55 const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); in AnalyzeStreamGaps()
58 for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) { in AnalyzeStreamGaps()
59 if (IsRtxSsrc(parsed_log, direction, stream.ssrc)) { in AnalyzeStreamGaps()
99 void TriageHelper::AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log, in AnalyzeTransmissionGaps() argument
123 const int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); in AnalyzeTransmissionGaps()
128 for (const auto& stream : parsed_log.rtp_packets_by_ssrc(direction)) { in AnalyzeTransmissionGaps()
150 for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) { in AnalyzeTransmissionGaps()
164 for (const auto& rtcp : parsed_log.outgoing_rtcp_packets()) { in AnalyzeTransmissionGaps()
184 void TriageHelper::AnalyzeLog(const ParsedRtcEventLog& parsed_log) { in AnalyzeLog() argument
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Dmain.cc257 webrtc::ParsedRtcEventLog parsed_log(header_extensions, in main() local
262 auto status = parsed_log.ParseFile(filename); in main()
273 if (!parsed_log.start_log_events().empty()) { in main()
274 config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() - in main()
275 parsed_log.start_log_events()[0].log_time(); in main()
278 config.begin_time_ = parsed_log.first_timestamp(); in main()
279 config.end_time_ = parsed_log.last_timestamp(); in main()
287 webrtc::EventLogAnalyzer analyzer(parsed_log, config); in main()
444 CreateAudioEncoderTargetBitrateGraph(parsed_log, config, plot); in main()
447 CreateAudioEncoderFrameLengthGraph(parsed_log, config, plot); in main()
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Danalyze_audio.cc27 void CreateAudioEncoderTargetBitrateGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderTargetBitrateGraph() argument
44 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderTargetBitrateGraph()
52 void CreateAudioEncoderFrameLengthGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderFrameLengthGraph() argument
69 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderFrameLengthGraph()
77 void CreateAudioEncoderPacketLossGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderPacketLossGraph() argument
94 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderPacketLossGraph()
103 void CreateAudioEncoderEnableFecGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderEnableFecGraph() argument
120 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderEnableFecGraph()
128 void CreateAudioEncoderEnableDtxGraph(const ParsedRtcEventLog& parsed_log, in CreateAudioEncoderEnableDtxGraph() argument
145 parsed_log.audio_network_adaptation_events(), &time_series); in CreateAudioEncoderEnableDtxGraph()
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Dalerts.h54 void AnalyzeLog(const ParsedRtcEventLog& parsed_log);
56 void AnalyzeStreamGaps(const ParsedRtcEventLog& parsed_log,
58 void AnalyzeTransmissionGaps(const ParsedRtcEventLog& parsed_log,
Danalyzer_common.h85 bool IsRtxSsrc(const ParsedRtcEventLog& parsed_log,
88 bool IsVideoSsrc(const ParsedRtcEventLog& parsed_log,
91 bool IsAudioSsrc(const ParsedRtcEventLog& parsed_log,
95 std::string GetStreamName(const ParsedRtcEventLog& parsed_log,
/external/webrtc/rtc_tools/rtc_event_log_to_text/
Dconverter.cc68 ParsedRtcEventLog parsed_log(handle_unconfigured_extensions, in Convert() local
71 auto status = parsed_log.ParseFile(inputfile); in Convert()
435 processor.AddEvents(parsed_log.audio_recv_configs(), in Convert()
437 processor.AddEvents(parsed_log.audio_send_configs(), in Convert()
439 processor.AddEvents(parsed_log.video_recv_configs(), in Convert()
441 processor.AddEvents(parsed_log.video_send_configs(), in Convert()
445 processor.AddEvents(parsed_log.start_log_events(), start_logging_handler); in Convert()
446 processor.AddEvents(parsed_log.stop_log_events(), stop_logging_handler); in Convert()
449 for (const auto& kv : parsed_log.audio_playout_events()) { in Convert()
452 processor.AddEvents(parsed_log.audio_network_adaptation_events(), in Convert()
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/external/webrtc/modules/audio_coding/neteq/tools/
Drtc_event_log_source.cc45 ParsedRtcEventLog parsed_log; in CreateFromFile() local
46 auto status = parsed_log.ParseFile(file_name); in CreateFromFile()
52 if (!source->Initialize(parsed_log, ssrc_filter)) { in CreateFromFile()
64 ParsedRtcEventLog parsed_log; in CreateFromString() local
65 auto status = parsed_log.ParseString(file_contents); in CreateFromString()
71 if (!source->Initialize(parsed_log, ssrc_filter)) { in CreateFromString()
99 bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log, in Initialize() argument
102 parsed_log.stop_log_events().empty() in Initialize()
104 : parsed_log.stop_log_events().front().log_time_us(); in Initialize()
135 for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) { in Initialize()
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Drtc_event_log_source.h59 bool Initialize(const ParsedRtcEventLog& parsed_log,
/external/webrtc/logging/rtc_event_log/
Drtc_event_log_unittest.cc565 ParsedRtcEventLog parsed_log; in ReadAndVerifyLog() local
568 ASSERT_TRUE(parsed_log.ParseString(it->second).ok()); in ReadAndVerifyLog()
571 auto& parsed_start_log_events = parsed_log.start_log_events(); in ReadAndVerifyLog()
576 auto& parsed_stop_log_events = parsed_log.stop_log_events(); in ReadAndVerifyLog()
580 auto& parsed_alr_state_events = parsed_log.alr_state_events(); in ReadAndVerifyLog()
586 auto& parsed_route_change_events = parsed_log.route_change_events(); in ReadAndVerifyLog()
593 const auto& parsed_audio_playout_map = parsed_log.audio_playout_events(); in ReadAndVerifyLog()
607 parsed_log.audio_network_adaptation_events(); in ReadAndVerifyLog()
615 auto& parsed_bwe_delay_updates = parsed_log.bwe_delay_updates(); in ReadAndVerifyLog()
622 auto& parsed_bwe_loss_updates = parsed_log.bwe_loss_updates(); in ReadAndVerifyLog()
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Drtc_event_log_parser.h927 const ParsedRtcEventLog& parsed_log); in RTC_POP_IGNORING_WUNDEF()
Drtc_event_log_parser.cc2484 const ParsedRtcEventLog& parsed_log) { in GetNetworkTrace() argument
2487 parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) { in GetNetworkTrace()
/external/webrtc/logging/rtc_event_log/encoder/
Drtc_event_log_encoder_unittest.cc87 const ParsedRtcEventLog* parsed_log,
142 RtcEventLogEncoderTest::GetRtpPacketsBySsrc(const ParsedRtcEventLog* parsed_log, in GetRtpPacketsBySsrc() argument
144 const auto& incoming_streams = parsed_log->incoming_rtp_packets_by_ssrc(); in GetRtpPacketsBySsrc()
155 RtcEventLogEncoderTest::GetRtpPacketsBySsrc(const ParsedRtcEventLog* parsed_log, in GetRtpPacketsBySsrc() argument
157 const auto& outgoing_streams = parsed_log->outgoing_rtp_packets_by_ssrc(); in GetRtpPacketsBySsrc()